Redirect SIP Trunk calls to FXO port

Hi,
This is the scenario. There are 3 branches, two of them are Cisco Call Manager Express and one of them is Elastix-based.
So, as the image explains, the three branches have SIP trunks fully operational. The branches are in different cities, so the numbers structure changes. In city A it begins with 2, in B begins with 3 and in C begins with 4. Every POTS number is a 7 digit number (2XXXXXX, 3XXXXXX, 4XXXXXX). And every user, in every branch, have a 4 digit number beginning with the city code (2XXX, 3XXX, 4XXX).
But, every time city A wants to make a call to a POTS number in city B, it goes across the A´s FXO line. So it charges a inter-city cost to the call.
The client wants that every time a city A user wants to call a POTS number in city B, goes over the SIP trunk to city B and use the FXO on the city B call manager.
I have made a pattern for city A. So, everytime the user dials 3XXXXXX, it does not use the city A´s FXO, but it goes to the branch in city B.
What do I have to do now in branch B´s Call Manager Express to redirect that call to a local FXO?
Thanks in advanced!
Regards
PS. There is  a diagram of the topology. Want to do what the red line is doing.

In this situation I would do an answer-address based on ANI so you are specifically identifying your site A and then just piggy back off the local FXO out.
So assuming you are sending just 4 digits over the SIP for each site:
Dial-peer voice X voip
answer-address "blah"
protocol sipv2
...(whatever else you need to configure in these dots)
At this point your CME at site B will take the call see that it is destined for a POTs line and it should send it out whatever local dial-peer you have setup for that site when they dial out to the PSTN locally.
EDIT:
Then again, you probably already have a general incoming dial-peer, the above design would just be specific for your site A and isn't really needed.

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    no service password-encryption
    service internal
    service compress-config
    service sequence-numbers
    hostname UC540
    boot-start-marker
    boot system flash:uc500-advipservicesk9-mz.151-2.T4
    boot-end-marker
    logging buffered 64000
    enable secret 5 $1$3CIf$.rXyHeJQrwd97X/f2dS0M1
    no aaa new-model
    clock timezone ZP4 4 0
    crypto pki token default removal timeout 0
    crypto pki trustpoint TP-self-signed-3558175224
    enrollment selfsigned
    subject-name cn=IOS-Self-Signed-Certificate-3558175224
    revocation-check none
    crypto pki certificate chain TP-self-signed-3558175224
    certificate self-signed 01 nvram:IOS-Self-Sig#3.cer
    dot11 syslog
    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 10.1.3.1 10.1.3.10
    ip dhcp pool phone
       network 10.1.3.0 255.255.255.0
       default-router 10.1.3.1
       option 150 ip 10.1.3.1
    ip name-server 213.42.20.20
    ip name-server 195.229.241.222
    ip inspect WAAS flush-timeout 10
    ip inspect name SDM_LOW cuseeme
    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
    stcapp supplementary-services
    port 0/0/0
      fallback-dn 301
    port 0/0/1
      fallback-dn 302
    port 0/0/2
      fallback-dn 303
    port 0/0/3
      fallback-dn 304
    trunk group ALL_FXO
    max-retry 5
    voice-class cause-code 1
    hunt-scheme longest-idle
    translation-profile outgoing PROFILE_ALL_FXO
    trunk group ALL_FX0
    voice call send-alert
    voice rtp send-recv
    voice service voip
    allow-connections h323 to h323
    allow-connections h323 to sip
    allow-connections sip to h323
    allow-connections sip to sip
    supplementary-service h450.12
    sip
      no update-callerid
    voice class codec 1
    codec preference 1 g711alaw
    codec preference 2 g711ulaw
    voice class dualtone-detect-params 1
    freq-max-deviation 50
    freq-max-power 0
    freq-min-power 13
    freq-power-twist 4
    cadence-variation 6
    voice class custom-cptone UAE-CUSTOM
    dualtone disconnect
      frequency 406
      cadence 398 344 237 527 400
    voice class custom-cptone CCAjointone
    dualtone conference
      frequency 600 900
      cadence 300 150 300 100 300 50
    voice class custom-cptone CCAleavetone
    dualtone conference
      frequency 400 800
      cadence 400 50 200 50 200 50
    voice class cause-code 1
    no-circuit
    voice register global
    voice hunt-group 1 parallel
    list 301,302,303
    timeout 24
    pilot 511
    voice translation-rule 4
    rule 15 // //
    voice translation-rule 1000
    rule 1 /.*/ //
    voice translation-rule 1111
    voice translation-rule 1112
    rule 1 /^9/ //
    rule 3 /^0/ //
    voice translation-rule 2222
    voice translation-rule 3265
    rule 1 /\(^..........$\)/ /9\1/
    rule 2 /\(^.........$\)/ /9\1/
    rule 15 /\(^ABCD$\)/ /ABCD\1/
    voice translation-profile CALLER_ID_TRANSLATION_PROFILE
    translate calling 1111
    voice translation-profile CallBlocking
    translate called 2222
    voice translation-profile INCOMING_CallerID_PROFILE
    translate calling 3265
    voice translation-profile OUTGOING_TRANSLATION_PROFILE
    translate called 1112
    voice translation-profile PROFILE_ALL_FXO
    translate calling 4
    voice translation-profile nondialable
    translate called 1000
    voice-card 0
    dspfarm
    dsp services dspfarm
    license udi pid UC540W-FXO-K9 sn FHK143074G6
    archive
    log config
      logging enable
      logging size 600
      hidekeys
    username cisco privilege 15 secret 5 $1$vjNa$OFKLhupqR8al6x2b8Xmcj/
    username adminac privilege 15 secret 5 $1$NDC.$PtD0y4YGIj5SqI1gghxWE1
    username Nick privilege 15 secret 5 $1$iAmL$tsg7Jf2TEND1NN.h8z2dy/
    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 $FW_OUTSIDE$
    ip address 192.168.101.2 255.255.255.252
    ip nat outside
    ip virtual-reassembly in
    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
    switchport voice vlan 100
    macro description cisco-phone
    spanning-tree portfast
    interface FastEthernet0/1/1
    switchport voice vlan 100
    macro description cisco-phone
    spanning-tree portfast
    interface FastEthernet0/1/2
    switchport voice vlan 100
    macro description cisco-phone
    spanning-tree portfast
    interface FastEthernet0/1/3
    switchport voice vlan 100
    macro description cisco-phone
    spanning-tree portfast
    interface FastEthernet0/1/4
    switchport voice vlan 100
    macro description cisco-phone
    spanning-tree portfast
    interface FastEthernet0/1/5
    switchport voice vlan 100
    macro description cisco-phone
    spanning-tree portfast
    interface FastEthernet0/1/6
    switchport voice vlan 100
    macro description cisco-phone
    spanning-tree portfast
    interface FastEthernet0/1/7
    switchport access vlan 20
    spanning-tree portfast
    interface FastEthernet0/1/8
    switchport access vlan 100
    macro description cisco-switch
    interface Dot11Radio0/5/0
    no ip address
    shutdown
    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
    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 Vlan20
    ip address 10.10.10.1 255.255.255.0
    interface Vlan100
    no ip address
    bridge-group 100
    bridge-group 100 spanning-disabled
    interface BVI1
    description $FW_INSIDE$
    no ip address
    ip nat inside
    ip virtual-reassembly in
    shutdown
    interface BVI100
    description $FW_INSIDE$
    ip address 10.1.3.1 255.255.255.0
    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 0.0.0.0 192.168.101.1
    ip route 10.1.10.1 255.255.255.255 Integrated-Service-Engine0/0
    logging esm config
    access-list 1 remark SDM_ACL Category=2
    access-list 1 permit 192.168.10.0 0.0.0.255
    access-list 1 permit 10.1.3.0 0.0.0.255
    access-list 1 permit 10.1.10.0 0.0.0.3
    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 10.1.3.0 0.0.0.255 eq 2000 any
    access-list 101 permit udp 10.1.3.0 0.0.0.255 eq 2000 any
    access-list 101 deny   ip 10.1.3.0 0.0.0.255 any
    access-list 101 deny   ip 192.168.10.0 0.0.0.255 any
    access-list 101 deny   ip 192.168.101.0 0.0.0.3 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 10.1.3.0 0.0.0.255 any
    access-list 102 deny   ip 192.168.101.0 0.0.0.3 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 102 permit ip 192.168.101.0 0.0.0.3 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 192.168.101.0 0.0.0.3 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 105 permit ip any any
    snmp-server community public RO
    tftp-server flash:/phones/521_524/cp524g-8-1-17.bin alias cp524g-8-1-17.bin
    tftp-server flash:/phones/5x5/spa5x5-7-1-3c.bin alias spa5x5-7-1-3c.bin
    tftp-server flash:/phones/525/spa525g-7-4-8.bin alias spa525g-7-4-8.bin
    control-plane
    bridge 1 route ip
    bridge 100 route ip
    voice-port 0/0/0
    cptone GB
    station-id name Cordless
    station-id number 329
    caller-id enable
    voice-port 0/0/1
    cptone AE
    caller-id enable
    voice-port 0/0/2
    cptone AE
    caller-id enable
    voice-port 0/0/3
    cptone AE
    caller-id enable
    voice-port 0/1/0
    trunk-group ALL_FX0 64
    translation-profile incoming INCOMING_CallerID_PROFILE
    supervisory disconnect dualtone mid-call
    supervisory custom-cptone UAE-CUSTOM
    input gain 14
    cptone GB
    connection plar opx 511
    impedance 600c
    description Configured by CCA 4FXO-0/1/0-Custom-BG
    bearer-cap Speech
    caller-id enable
    voice-port 0/1/1
    trunk-group ALL_FX0 64
    translation-profile incoming INCOMING_CallerID_PROFILE
    supervisory disconnect dualtone mid-call
    supervisory custom-cptone UAE-CUSTOM
    input gain 14
    cptone GB
    connection plar opx 511
    impedance 600c
    description Configured by CCA 4 FXO-0/1/1-Custom-BG
    bearer-cap Speech
    caller-id enable
    voice-port 0/1/2
    trunk-group ALL_FX0 64
    translation-profile incoming INCOMING_CallerID_PROFILE
    supervisory disconnect dualtone mid-call
    supervisory custom-cptone UAE-CUSTOM
    supervisory dualtone-detect-params 1
    input gain 14
    cptone GB
    connection plar opx 511
    impedance 600c
    description Configured by CCA 4 FXO-0/1/2-Custom-BG
    bearer-cap Speech
    caller-id enable
    voice-port 0/1/3
    trunk-group ALL_FX0 64
    translation-profile incoming INCOMING_CallerID_PROFILE
    supervisory disconnect dualtone mid-call
    supervisory custom-cptone UAE-CUSTOM
    input gain 14
    cptone GB
    connection plar opx 511
    impedance 600c
    description Configured by CCA 4 FXO-0/1/3-Custom-BG
    bearer-cap Speech
    caller-id enable
    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 10.1.3.1 identifier 1 version 4.0
    sccp
    sccp ccm group 1
    associate ccm 1 priority 1
    associate profile 1 register confprof1
    dspfarm profile 1 conference 
    description DO NOT MODIFY, active CCA conference profile - CCA2.0 codec729
    codec g711ulaw
    codec g711alaw
    codec g729ar8
    codec g729abr8
    codec g729r8
    codec g729br8
    maximum sessions 2
    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
    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 50 pots
    description ** incoming dial peer **
    incoming called-number ^AAAA$
    port 0/1/0
    dial-peer voice 51 pots
    description ** incoming dial peer **
    incoming called-number ^AAAA$
    port 0/1/1
    dial-peer voice 52 pots
    description ** incoming dial peer **
    incoming called-number ^AAAA$
    port 0/1/2
    dial-peer voice 53 pots
    description ** incoming dial peer **
    incoming called-number ^AAAA$
    port 0/1/3
    dial-peer voice 54 pots
    description ** FXO pots dial-peer **
    destination-pattern A0
    port 0/1/0
    no sip-register
    dial-peer voice 55 pots
    description ** FXO pots dial-peer **
    destination-pattern A1
    port 0/1/1
    no sip-register
    dial-peer voice 56 pots
    description ** FXO pots dial-peer **
    destination-pattern A2
    port 0/1/2
    no sip-register
    dial-peer voice 2000 voip
    description ** cue voicemail pilot number **
    destination-pattern 388
    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 6 pots
    description "catch all dial peer for BRI/PRI"
    translation-profile incoming nondialable
    incoming called-number .%
    direct-inward-dial
    dial-peer voice 57 pots
    description ** FXO pots dial-peer **
    destination-pattern A3
    port 0/1/3
    no sip-register
    dial-peer voice 69 pots
    destination-pattern 329
    port 0/0/0
    dial-peer voice 300 pots
    trunkgroup ALL_FX0
    description Local Numbers
    destination-pattern 9T
    forward-digits 9
    dial-peer voice 301 voip
    destination-pattern 2..
    session target ipv4:192.168.201.2
    dial-peer voice 303 pots
    trunkgroup ALL_FXO
    trunkgroup ALL_FX0
    description **InternationalCall**
    destination-pattern 88T
    dial-peer voice 304 pots
    trunkgroup ALL_FX0
    description *EM1*
    destination-pattern 9[1-9]T
    forward-digits 3
    dial-peer voice 302 pots
    trunkgroup ALL_FX0
    description **Mobiles**
    destination-pattern 9.[0-9].[0-9]......
    dial-peer voice 305 pots
    trunkgroup ALL_FX0
    description **800-**
    destination-pattern 9[0-9][0-9][0-9]T
    no dial-peer outbound status-check pots
    telephony-service
    sdspfarm conference mute-on 111 mute-off 222
    sdspfarm units 5
    sdspfarm tag 1 confprof1
    conference hardware
    video
    fxo hook-flash
    max-ephones 40
    max-dn 300
    ip source-address 10.1.3.1 port 2000
    max-redirect 20
    auto assign 1 to 1 type bri
    calling-number initiator
    service phone videoCapability 1
    service phone webAccess 0
    service dnis overlay
    service dnis dir-lookup
    timeouts interdigit 5
    system message American Center
    url services http://10.1.10.1/voiceview/common/login.do
    url authentication http://10.1.10.2/CCMCIP/authenticate.asp 
    load 521G-524G cp524g-8-1-17
    load 525G spa525g-7-4-8
    load 501G spa5x5-7-1-3c
    load 502G spa5x5-7-1-3c
    load 504G spa5x5-7-1-3c
    load 508G spa5x5-7-1-3c
    load 509G spa5x5-7-1-3c
    time-zone 35
    date-format dd-mm-yy
    voicemail 388
    max-conferences 8 gain -6
    call-forward pattern .T
    call-forward system redirecting-expanded
    hunt-group logout HLog
    moh MOH2.wav
    multicast moh 239.10.16.16 port 2000
    web admin system name cisco secret 5 $1$iDgA$MKNi2RWfsO0KjuC82kgLJ1
    dn-webedit
    time-webedit
    transfer-system full-consult dss
    transfer-pattern 9.T
    transfer-pattern .T
    secondary-dialtone 9
    fac standard
    create cnf-files version-stamp 7960 Aug 29 2012 12:00:04
    line con 0
    privilege level 15
    logging synchronous
    no modem enable
    line aux 0
    line 2
    no activation-character
    no exec
    transport preferred none
    transport input all
    line vty 0 4
    exec-timeout 0 0
    logging synchronous
    login local
    transport input all
    line vty 5 100
    login local
    transport input all
    ntp master
    end
    Some of the output are not shown becaus it is to long I have attach the  whole config for reference and any advice on how could I optimize and  resolve my issues is greatly appreciated. Thanks

    Nicolo - First off this stuff gets crazy sometimes.  No worries about the exam.  Sometimes when FXO ports go crazy it is due to battery reversal.  If you go to the FXO port settings try turning battery reversal on and or off... depending on its current setting.  See if that helps. 
    As for the 525s not registering..  These are inside the network correct?  Are you connecting one directly to the UC500 with a Cat5E or Cat6 patch cable and the same thing happens?  Does the MAC address on the phone match a MAC address under the EPHONE settings? 
    If you telnet into the UC500 can you execute a "dir" command at the CLI prompt and "CD" (change directory) into the phones folder and then the spa525g folder?  Do files exist in there? 
    Also I only see an IP address under BVI100?  This is the voice side of things what happened to the IP address under BVI1 (Data VLAN).  Can you give us some information about the internal network?  Cna you PING this phone system from the network?  What IP address does it have?

  • Transfer called number from FSX port SPA8000 to FXO port analog PBX.

    Hello all!
    I have CUCM 8.6 which connecting with SPA8000 by SIP trunk.
    FXS port SPA8000 connect to FXO port analog PBX.
    When I call from CUCM to analog PBX through SPA8000, SPA8000 does not pass called number in line.
    Anybody have ideas?

    Here is what I would recomend, I have set this up at several installations and it works well. The only difference would be that the ones that I have setup had more than 64K between sites but that is still doable so long as you keep the traffic on that link well regulated.
    Anyhow. At each site install a VIC-2FXO and a VIC-2FXS into your 1760. You can use the higer density modules if you need more paths. Tie the FXOs to analog station ports on your PBX and the FXSs to analog CO ports. This may seem backwards but it is correct. Cisco's FXS is expecting to provide dialtone and the analog CO ports on the PBX are expecting to receive dial tone. Depending on what you are using for call processing ITS/CCME/CCM configure your gateway and your route pattern, etc. You will also need to be able to somehow access those fxs ports for outbound calls. You could have direct line access buttons on your phones and instruct users that when calling the remote site the must use lines 7 and 8 as an example. You could also setup a line pool code for these new lines.
    Now when site A wants to call Site B, they dial whatever you wish them to dial. You do whatever translations you may wish prior to sending the call. Here is how the call flow works. User at site A accesses the line either by hitting a line button or dialing the line pool code. This grabs the analog line tied to one of the FXs ports on the router. The call is then sent over the 64K link. The router at Site B then uses one of the FXO ports to pass the call to an analog station port on the other PBX. To the PBX at site B it will appear as though someone pickedu picked up an analog station and dialed a local extension.
    Hope this helps. If you need any further clarification on this let me know and I will try to clear it up further for you.

  • UC520 SIP trunk unable to make outgoing calls, incoming calls are ok

    I have an new SIP trunk set on an UC520 and the incoming calls are ok, but the outgoing calls are getting an busy tone(not working).
    The bellow trace is showing that the cause is "No route to destination (3) ". The question is this route has to do with the firewall(ip routing) or with the voice translation rules?
    001866: //3439/91242E51926F/SIP/Call/sipSPICallInfo:
    The Call Setup Information is:
    Call Control Block (CCB) : 0x843CE50C
    State of The Call        : STATE_DEAD
    TCP Sockets Used         : YES
    Calling Number           : 0777777777 <- main sip number
    Called Number            : 0888888888 <- called number
    Source IP Address (Sig  ): 0.0.0.0
    Destn SIP Req Addr:Port  : 0.0.0.0:0
    Destn SIP Resp Addr:Port : 0.0.0.0:0
    Destination Name         :
    001867: //3439/91242E51926F/SIP/Call/sipSPICallInfo:
    Disconnect Cause (CC)    : 3
    Disconnect Cause (SIP)   : 200

    Hi Emil,
    I've added bind control and media interface but outgoing calls are the same blocked, strange thing is that the cause is still no route to destination (3)
    but
    UC_520#show sip-ua status
    SIP User Agent Status
    SIP User Agent for UDP : ENABLED
    SIP User Agent for TCP : ENABLED
    SIP User Agent for TLS over TCP : ENABLED
    SIP User Agent bind status(signaling): ENABLED 10.10.10.5 <- fa0/0 IP address
    SIP User Agent bind status(media): ENABLED 10.10.10.5 <- fa0/0 IP address
    SIP early-media for 180 responses with SDP: ENABLED
    SIP max-forwards : 70
    SIP DNS SRV version: 2 (rfc 2782)
    NAT Settings for the SIP-UA
    Role in SDP: NONE
    Check media source packets: DISABLED
    Maximum duration for a telephone-event in NOTIFYs: 2000 ms
    SIP support for ISDN SUSPEND/RESUME: ENABLED
    Redirection (3xx) message handling: ENABLED
    Reason Header will override Response/Request Codes: DISABLED
    Out-of-dialog Refer: DISABLED
    Presence support is DISABLED
    SDP application configuration:
     Version line (v=) required
     Owner line (o=) required
     Timespec line (t=) required
     Media supported: audio image
     Network types supported: IN
     Address types supported: IP4
     Transport types supported: RTP/AVP udptl

  • SIP Trunk not accepting inbound calls

    I have a CME setup using Engin as a SIP provider
    I am able to dial out with no issue, however my inbound calls do not work, they divert to the Engin voicemail
    My SIP registration is OK and the number is configured as the primary DN on one of my phones
    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
    I am seeing a couple of disconnect cause codes that I cant seem to find any relavent information on in the CCSIP debugs
    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

  • Unable to place call on calls on hold - SIP Trunk from CUCM to CUBE and from CUBE to ISTP

    Hi Cisco Community,
    I have a SIP Trunk setup between the CUCM and CUBE and another SIP Dial Peers from the CUBE to the ITSP. All incoming/outgoing calls, DTMF-Relay works well except one thing which is the ability to hold the call.
    On the SIP Trunk from the CUCM to the CUBE, I did not select “MTP” because when I do so, I am forced to select my preferred MTP codec which when selected G.729/G.729a, all my outgoing calls goes out using G.729r8. This codec works well for most of the calls until the ITSP replies back with G.729br8. When this condition occur, my call simply fails (this is very intermittent and only some random numbers).
    That said, I have some issues with DTMF Relay when I select MTP on the SIP Trunk. DTMF Relay only works if the call is G.729r8 all the way from the CUCM to the ITSP. If the ITSP replies back with G.729br8, the call might established but will simply be “voice-only”.
    The current setup is no MTP is selected and everything is working perfectly. I am happy with that until I place a call on hold, which when I do so, the call immediately terminate. Could you please help me understand why?
    I have all media resources configured such as G.729r8 MTP, G.729br8 MTP, G.711u MTP, Transcoding with all codecs, etc. All MRG and MRGL are configured on all devices and SIP Trunks.
    Below is an example of a call that is connected with the current setup:
    Note:
    IP: 10.18.81.2 (CUBE)
    IP: 10.18.81.11 (CUCM SUB)
    IP: 10.111.111.254 (ITSP SBC)
    PM-HO-VG-01#
    PM-HO-VG-01#
    Nov 30 11:44:29.938: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received: 
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM9.1
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 101 INVITE
    Expires: 180
    Allow-Events: presence, kpml
    Supported: X-cisco-srtp-fallback,X-cisco-original-called
    Call-Info: <sip:10.18.81.11:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
    Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
    Session-Expires:  1800
    Contact: <sip:[email protected]:5060>
    Max-Forwards: 70
    Content-Length: 0
    Nov 30 11:44:29.942: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Content-Length: 0
    Nov 30 11:44:29.946: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
    User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Timestamp: 1417347869
    Contact: <sip:[email protected]:5060>
    Call-Info: <sip:10.18.81.2:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
    Expires: 180
    Allow-Events: telephone-event
    Max-Forwards: 69
    Session-Expires:  1800
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 301
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7676 6958 IN IP4 10.18.81.2
    s=SIP Call
    c=I
    PM-HO-VG-01#N IP4 10.18.81.2
    t=0 0
    m=audio 22256 RTP/AVP 18 0 8 101
    c=IN IP4 10.18.81.2
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    Nov 30 11:44:29.950: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Timestamp: 1417347869
    Nov 30 11:44:30.658: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 180 Session Progress
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Timestamp: 1417347869
    Supported: 
    Contact: <sip:[email protected]:5060;transport=udp>
    Session: Media
    Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
    X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
    Content-Type: application/sdp
    Content-Length: 355
    v=0
    o=BroadWorks 316169737 1 IN IP4 10.111.111.254
    s=-
    c=IN IP4 10.111.111.254
    t=0 0
    m=audio 20074 RTP/AVP 18 101 100
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:100 X-NSE/8000
    a=fmtp:100 200-202
    a=X-sqn:0
    a=X-cap: 1 audio RTP/AVP 100
    a=X-cpar: a=rtpmap:100 X-NSE/8000
    a=X-cpar: a=fmtp:100 200-202
    a=X-cap: 2 image udptl t38
    Nov 30 11:44:30.662: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 183 Session Progress
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Contact: <sip:[email protected]:5060>
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 289
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7965 2747 IN IP4 10.18.81.2
    s=SIP Call
    c=IN IP4 10.18.81.2
    t=0 0
    m=audio 22350 RTP/AVP 18 101 19
    c=IN IP4 10.18.81.2
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:19 CN/8000
    a=ptime:20
    Nov 30 11:44:31.226: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 180 Session Progress
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Timestamp: 1417347869
    Supported: 
    Contact: <sip:[email protected]:5060;transport=udp>
    Session: Media
    Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
    X-BroadWorks-Correlation-Info: bbf9
    PM-HO-VG-01#4839-a234-4237-95e6-a7037322f0f4
    Content-Type: application/sdp
    Content-Length: 355
    v=0
    o=BroadWorks 316169737 1 IN IP4 10.111.111.254
    s=-
    c=IN IP4 10.111.111.254
    t=0 0
    m=audio 20074 RTP/AVP 18 101 100
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:100 X-NSE/8000
    a=fmtp:100 200-202
    a=X-sqn:0
    a=X-cap: 1 audio RTP/AVP 100
    a=X-cpar: a=rtpmap:100 X-NSE/8000
    a=X-cpar: a=fmtp:100 200-202
    a=X-cap: 2 image udptl t38
    Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Timestamp: 1417347869
    Supported: 
    Contact: <sip:[email protected]:5060;transport=udp>
    Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
    Accept: application/media_control+xml,application/sdp,application/xml
    X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
    Content-Type: application/sdp
    Content-Length: 355
    v=0
    o=BroadWorks 316169737 1 IN IP4 10.111.111.254
    s=-
    c=IN IP4 10.111.111.254
    t=0 0
    m=audio 20074 RTP/AVP 18 101 100
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:100 X-NSE/8000
    a=fmtp:100 200-202
    a=X-sqn:0
    a=X-cap: 1 audio RTP/AVP 100
    a=X-cpar: a=rtpmap:100 X-NSE/8000
    a=X-cpar: a=fmtp:100 200-202
    a=X-cap: 2 image udptl t38
    Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    The Call Setup Information is:
    Call Control Block (CCB) : 0x0x3D7B1458
    State of The Call        : STATE_ACTIVE
    TCP Sockets Used         : NO
    Calling Number           : 27218091323
    Called Number            : 0862000000
    Source IP Address (Sig  ): 10.18.81.2
    Destn SIP Req Addr:Port  : 10.111.111.254:5060
    Destn SIP Resp Addr:Port : 10.111.111.254:5060
    Destination Name         : 10.111.111.254
    Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo: 
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g729br8
    Negotiated Codec Bytes   : 20
    Nego. Codec payload      : 18 (tx), 18 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
    Source IP Address (Media): 10.18.81.2
    Source IP Port    (Media): 22256
    Destn  IP Address (Media): 10.111.111.254
    Destn  IP Port    (Media): 20074
    Orig Destn IP Address:Port (Media): [ - ]:0
    Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    ACK sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC81D00
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: telephone-event
    Content-Length: 0
    Nov 30 11:44:31.634: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Contact: <sip:[email protected]:5060>
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Supported: timer
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 289
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7965 2747 IN IP4 10.18.81.2
    s=SIP Call
    c=IN IP4 10.18.81.2
    t=0 0
    m=audio 22350 RTP/AVP 18 101 19
    c=IN IP4 10.18.81.2
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:19 CN/8000
    a=ptime:20
    Nov 30 11:44:31.726: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received: 
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72075e3a02c1
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: presence, kpml
    Content-Type: application/sdp
    Content-Length: 236
    v=0
    o=CiscoSystemsCCM-SIP 9082578 1 IN IP4 10.18.81.11
    s=SIP Call
    c=IN IP4 10.18.80.40
    b=TIAS:8000
    b=AS:8
    t=0 0
    m=audio 21928 RTP/AVP 18 101
    a=rtpmap:18 G729/8000
    a=ptime:20
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    The Call Setup Information is:
    Call Control Block (CCB) : 0x0x3D816D70
    State of The Call        : STATE_ACTIVE
    TCP Sockets Used         : NO
    Calling Number           : 0218091323
    Called Number            : 0862000000
    Source IP Address (Sig  ): 10.18.81.2
    Destn SIP Req Addr:Port  : 10.18.81.11:5060
    Destn SIP Resp Addr:Port : 10.18.81.11:5060
    Destination Name         : 10.18.81.11
    Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo: 
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g729br8
    Negotiated Codec Bytes   : 20
    Nego. Codec payload      : 18 (tx), 18 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
    Source IP Address (Media): 10.18.81.2
    Source IP Port    (Media): 22350
    Destn  IP Address (Media): 10.18.80.40
    Destn  IP Port    (Media): 21928
    Orig Destn IP Address:Port (Media): [ - ]:0
    Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    The Call Setup Information is:
    Call Control Block (CCB) : 0x0x3D816D70
    State of The Call        : STATE_ACTIVE
    TCP Sockets Used         : NO
    Calling Number           : 0218091323
    Called Number            : 0862000000
    Source IP Address (Sig  ): 10.18.81.2
    Destn SIP Req Addr:Port  : 10.18.81.11:5060
    Destn SIP Resp Addr:Port : 10.18.81.11:5060
    Destination Name         : 10.18.81.11
    PM-HO-VG-01#
    Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo: 
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g729br8
    Negotiated Codec Bytes   : 20
    Nego. Codec payload      : 18 (tx), 18 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
    Source IP Address (Media): 10.18.81.2
    Source IP Port    (Media): 22350
    Destn  IP Address (Media): 10.18.80.40
    Destn  IP Port    (Media): 21928
    Orig Destn IP Address:Port (Media): [ - ]:0
    PM-HO-VG-01#sh sip
    PM-HO-VG-01#sh sip-ua call
    PM-HO-VG-01#sh sip-ua calls 
    Total SIP call legs:2, User Agent Client:1, User Agent Server:1
    SIP UAC CALL INFO
    Call 1
    SIP Call ID                : [email protected]
       State of the call       : STATE_ACTIVE (7)
       Substate of the call    : SUBSTATE_NONE (0)
       Calling Number          : 27218091323
       Called Number           : 0862000000
       Bit Flags               : 0xC04018 0x10000100 0x0
       CC Call ID              : 64511
       Source IP Address (Sig ): 10.18.81.2
       Destn SIP Req Addr:Port : [10.111.111.254]:5060
       Destn SIP Resp Addr:Port: [10.111.111.254]:5060
       Destination Name        : 10.111.111.254
       Number of Media Streams : 1
       Number of Active Streams: 1
       RTP Fork Object         : 0x0
       Media Mode              : flow-through
       Media Stream 1
         State of the stream      : STREAM_ACTIVE
         Stream Call ID           : 64511
         Stream Type              : voice+dtmf (0)
         Stream Media Addr Type   : 1
         Negotiated Codec         : g729br8 (20 bytes)
         Codec Payload Type       : 18 
         Negotiated Dtmf-relay    : rtp-nte
         Dtmf-relay Payload Type  : 101
         QoS ID                   : -1
         Local QoS Strength       : BestEffort
         Negotiated QoS Strength  : BestEffort
         Negotiated QoS Direction : None
         Local QoS Status         : None
         Media Source IP Addr:Port: [10.18.81.2]:22256
         Media Dest IP Addr:Port  : [10.111.111.254]:20074
    Options-Ping    ENABLED:NO    ACTIVE:NO
       Number of SIP User Agent Client(UAC) calls: 1
    SIP UAS CALL INFO
    Call 1
    SIP Call ID                : [email protected]
       State of the call       : STATE_ACTIVE (7)
       Substate of the call    : SUBSTATE_NONE (0)
       Calling Number          : 0218091323
       Called Number           : 0862000000
       Bit Flags               : 0xC0401E 0x10000100 0x80004
       CC Call ID              : 64510
       Source IP Address (Sig ): 10.18.81.2
       Destn SIP Req Addr:Port : [10.18.81.11]:5060
       Destn SIP Resp Addr:Port: [10.18.81.11]:5060
       Destination Name        : 10.18.81.11
       Number of Media Streams : 1
       Number of Active Streams: 1
       RTP Fork Object         : 0x0
       Media Mode              : flow-through
       Media Stream 1
         State of the stream      : STREAM_ACTIVE
         Stream Call ID           : 64510
         Stream Type              : voice+dtmf (1)
         Stream Media Addr Type   : 1
         Negotiated Codec         : g729br8 (20 bytes)
         Codec Payload Type       : 18 
         Negotiated Dtmf-relay    : rtp-nte
         Dtmf-relay Payload Type  : 101
         QoS ID                   : -1
         Local QoS Strength       : BestEffort
         Negotiated QoS Strength  : BestEffort
         Negotiated QoS Direction : None
         Local QoS Status         : None
         Media Source IP Addr:Port: [10.18.81.2]:22350
         Media Dest IP Addr:Port  : [10.18.80.40]:21928
    Options-Ping    ENABLED:NO    ACTIVE:NO
       Number of SIP User Agent Server(UAS) calls: 1
    PM-HO-VG-01#
    PM-HO-VG-01#
    PM-HO-VG-01#
    As you can see, the call is connected and everything is working perfectly. When I press the hold button, here is what I get:
    NOTE: I have # debug ccsip messages and #debug ccsip calls (running)
    PM-HO-VG-01#
    PM-HO-VG-01#
    Nov 30 11:44:49.210: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received: 
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM9.1
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 102 INVITE
    Max-Forwards: 70
    Expires: 180
    Allow-Events: presence
    Supported: X-cisco-srtp-fallback
    Supported: Geolocation
    Contact: <sip:[email protected]:5060>
    Content-Type: application/sdp
    Content-Length: 244
    v=0
    o=CiscoSystemsCCM-SIP 9082578 2 IN IP4 10.18.81.11
    s=SIP Call
    c=IN IP4 0.0.0.0
    b=TIAS:8000
    b=AS:8
    t=0 0
    m=audio 21928 RTP/AVP 18 101
    a=rtpmap:18 G729/8000
    a=ptime:20
    a=inactive
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    Nov 30 11:44:49.218: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    CSeq: 102 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Content-Length: 0
    Nov 30 11:44:49.218: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    INVITE sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC9241
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
    User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 102 INVITE
    Max-Forwards: 70
    Timestamp: 1417347889
    Contact: <sip:[email protected]:5060>
    Call-Info: <sip:10.18.81.2:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
    Expires: 180
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Length: 271
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7676 6959 IN IP4 10.18.81.2
    s=SIP Call
    c=IN IP4 0.0.0.0
    t=0 0
    m=audio 22256 RTP/AVP 18 101
    c=IN IP4 0.0.0.0
    a=inactive
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC9241
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Call-ID: [email protected]
    CSeq: 102 INVITE
    Timestamp: 1417347889
    Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
    Supported: 
    Accept: application/media_control+xml,application/sdp,application/xml
    Contact: <sip:[email protected]:5060;transport=udp>
    X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
    Content-Type: application/sdp
    Content-Length: 360
    v=0
    o=BroadWorks 316169737 2 IN IP4 10.111.111.254
    s=-
    c=IN IP4 0.0.0.0
    t=0 0
    m=audio 20074 RTP/AVP 18 101 100
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:100 X-NSE/8000
    a=fmtp:100 200-202
    a=X-sqn:0
    a=X-cap: 1 audio RTP/AVP 100
    a=X-cpar: a=rtpmap:100 X-NSE/8000
    a=X-cpar: a=fmtp:100 200-202
    a=X-cap: 2 image udptl t38
    a=inactive
    Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    The Call Setup Information is:
    Call Control Block (CCB) : 0x0x3D7B1458
    State of The Call        : STATE_ACTIVE
    TCP Sockets Used         : NO
    Calling Number           : 27218091323
    Called Number            : 0862000000
    Source IP Address (Sig  ): 10.18.81.2
    Destn SIP Req Addr:Port  : 10.111.111.254:5060
    Destn SIP Resp Addr:Port : 10.111.111.254:5060
    Destination Name         : 10.111.111.254
    Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo: 
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g729br8
    Negotiated Codec Bytes   : 20
    Nego. Codec payload      : 18 (tx), 18 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
    Source IP Address (Media): 10.18.81.2
    Source IP Port    (Media): 22256
    Destn  IP Address (Media): 0.0.0.0
    Destn  IP Port    (Media): 20074
    Orig Destn IP Address:Port (Media): [ - ]:0
    Nov 30 11:44:49.282: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    ACK sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECA2633
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 102 ACK
    Allow-Events: telephone-event
    Content-Length: 0
    Nov 30 11:44:49.282: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    CSeq: 102 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Contact: <sip:[email protected]:5060>
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Supported: timer
    Content-Type: application/sdp
    Content-Length: 271
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7965 2748 IN IP4 10.18.81.2
    s=SIP Call
    c=IN IP4 0.0.0.0
    t=0 0
    m=audio 22350 RTP/AVP 18 101
    c=IN IP4 0.0.0.0
    a=inactive
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    Nov 30 11:44:49.282: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received: 
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72094953dfea
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 102 ACK
    Allow-Events: presence
    Content-Length: 0
    Nov 30 11:44:49.290: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received: 
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM9.1
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 103 INVITE
    Max-Forwards: 70
    Expires: 180
    Allow-Events: presence
    Supported: X-cisco-srtp-fallback
    Supported: Geolocation
    Contact: <sip:[email protected]:5060>
    Content-Length: 0
    Nov 30 11:44:49.294: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    CSeq: 103 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Content-Length: 0
    Nov 30 11:44:49.294: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    INVITE sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECB16F3
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
    User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 103 INVITE
    Max-Forwards: 70
    Timestamp: 1417347889
    Contact: <sip:[email protected]:5060>
    Expires: 180
    Allow-Events: telephone-event
    Content-Length: 0
    Nov 30 11:44:49.338: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECB16F3
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Call-ID: [email protected]
    CSeq: 103 INVITE
    Timestamp: 1417347889
    Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
    Supported: 
    Accept: application/media_control+xml,application/sdp,application/xml
    Contact: <sip:[email protected]:5060;transport=udp>
    Content-Type: application/sdp
    Content-Length: 306
    v=0
    o=BroadWorks 316169737 3 IN IP4 10.111.111.254
    s=-
    c=IN IP4 10.111.111.254
    t=0 0
    m=audio 20074 RTP/AVP 18 101
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=X-sqn:0
    a=X-cap: 1 audio RTP/AVP 100
    a=X-cpar: a=rtpmap:100 X-NSE/8000
    a=X-cpar: a=fmtp:100 200-202
    a=X-cap: 2 image udptl t38
    Nov 30 11:44:49.342: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 2
    PM-HO-VG-01#00 OK
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    CSeq: 103 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Contact: <sip:[email protected]:5060>
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Supported: timer
    Content-Type: application/sdp
    Content-Length: 289
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7965 2749 IN IP4 10.18.81.2
    s=SIP Call
    c=IN IP4 10.18.81.2
    t=0 0
    m=audio 22350 RTP/AVP 18 101 19
    c=IN IP4 10.18.81.2
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:19 CN/8000
    a=ptime:20
    Nov 30 11:44:49.350: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received: 
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720b594cd517
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 103 ACK
    Allow-Events: presence
    Content-Type: application/sdp
    Content-Length: 213
    v=0
    o=CiscoSystemsCCM-SIP 9082578 3 IN IP4 10.18.81.11
    s=SIP Call
    c=IN IP4 10.18.81.10
    t=0 0
    m=audio 4000 RTP/AVP 18
    a=X-cisco-media:umoh
    a=rtpmap:18 G729/8000
    a=ptime:20
    a=fmtp:18 annexb=no
    a=sendonly
    Nov 30 11:44:49.354: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    BYE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECC55
    From: <sip:[email protected]>;tag=3C365010-1E42
    To: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
    Max-Forwards: 70
    Timestamp: 1417347889
    CSeq: 101 BYE
    Reason: Q.850;cause=86
    P-RTP-Stat: PS=874,OS=17480,PR=872,OR=17440,PL=0,JI=0,LA=0,DU=17
    Content-Length: 0
    Nov 30 11:44:49.354: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    BYE sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECD1ECD
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
    Max-Forwards: 70
    Timestamp: 1417347889
    CSeq: 104 BYE
    Reason: Q.850;cause=65
    P-RTP-Stat: PS=872,OS=17440,PR=952,OR=19040,PL=0,JI=0,LA=0,DU=17
    Content-Length: 0
    Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 200 Race Condition
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECD1ECD
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Call-ID: [email protected]
    Timestamp: 1417347889
    CSeq: 104 BYE
    Content-Length: 0
    Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    The Call Setup Information is:
    Call Control Block (CCB) : 0x0x3D7B1458
    State of The Call        : STATE_DEAD
    TCP Sockets Used         : NO
    Calling Number           : 27218091323
    Called Number            : 0862000000
    Source IP Address (Sig  ): 10.18.81.2
    Destn SIP Req Addr:Port  : 10.111.111.254:5060
    Destn SIP Resp Addr:Port : 10.111.111.254:5060
    Destination Name         : 10.111.111.254
    Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo: 
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g729br8
    Negotiated Codec Bytes   : 20
    Nego. Codec payload      : 18 (tx), 18 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
    Source IP Address (Media): 10.18.81.2
    Source IP Port    (Media): 22256
    Destn  IP Address (Media): 10.111.111.254
    Destn  IP Port    (Media): 20074
    Orig Destn IP Address:Port (Media): [ - ]:0
    Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    Disconnect Cause (CC)    : 65
    Disconnect Cause (SIP)   : 200
    Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECC55
    From: <sip:[email protected]>;tag=3C365010-1E42
    To: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    CSeq: 101 BYE
    Content-Length: 0
    Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    The Call Setup Information is:
    Call Control Block (CCB) : 0x0x3D816D70
    State of The Call        : STATE_DEAD
    TCP Sockets Used         : NO
    Calling Number           : 0218091323
    Called Number            : 0862000000
    Source IP Address (Sig  ): 10.18.81.2
    Destn SIP Req Addr:Port  : 10.18.81.11:5060
    Destn SIP Resp Addr:Port : 10.18.81.11:5060
    Destination Name         : 10.18.81.11
    Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo: 
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g729br8
    Negotiated Codec Bytes   : 20
    Nego. Codec payload      : 18 (tx), 18 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
    Source IP Address (Media): 10.18.81.2
    Source IP Port    (Media): 22350
    Destn  IP Address (Media): 0.0.0.0
    Destn  IP Port    (Media): 21928
    Orig Destn IP Address:Port (Media): [ - ]:0
    Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    Disconnect Cause (CC)    : 86
    Disconnect Cause (SIP)   : 200
    PM-HO-VG-01#

    Hi Manish,
    Again, excellent feedback. Much appreciated.
    I will try the commands suggested above and see if I can get DTMF to work correctly while interworking H.323 and SIP.
    But my ultimate goal is to have SIP all way from the CUCM to the CUBE and from the CUBE to the ITSP.
    If I enable SIP Early Offer with MTP on the CUCM going to the CUBE, all SIP Invite sent from the CUCM to the CUBE uses G.729r8 as the codec and once the call is established using G.729r8 should the ITSP reply with that codec, the call succeed and I am able to see an active MTP session using G.729 when I issue the command # show sccp connections.
    One thing that I saw is that my ITSP love so much sending G.729br8 most of the times, so even if using SIP EO with MTP on the SIP Trunk to the CUBE, when I sent my INVITE out from the CUCM to the CUBE using G.729r8, specially on call center numbers such as 0800 numbers, you will see that the call established but the codec being negotiated is G.729br8 which is voice only (missing DTMF).
    I will be doing some intensive test again later on this week and will send the logs. 
    Here is my question to both of you:
    Which is the best way of having a proper SIP to SIP setup all the way that will not pose any problem?
    Do I have to enable Early Offer on the SIP Profile used by the CUBE SIP Trunk or should I use the normal Standard SIP Profile? Do I need to enable MTP on my CUBE SIP Trunk or not?
    From the CUBE point of view, I have a voice class codec that support G.729r8 or G.729br8 and the DTMF Relay method supported by the ITSP is RFC 2833.
    I will send more logs for each scenario. I think that we are getting close to the resolution of this problem.
    Thanks again for your support fellows.

  • Incorrect Caller ID on calls from outside line via FXO port.

       Have a public phone line connected to my CUCME 2801 router VIC2-2FXO card. All inbound calls are passed to DN-5001 (group number). Can receive and send calls without a problem, but incoming calls all show "911" for caller ID. Think this is simply an issue with the out bound dial-peer, of which the lowest numbered out bound dial-peer is for 911 services. Not sure how to correct this so inbound calls show the proper caller ID?
        Below is a copy of my CUCME show run output from the FXO port config thru all the dial-peers. Any pointers is greatly appreciated.
        Thanks.
               Kirk E.
    voice-port 0/0/0
    connection plar opx immediate 5001
    voice-port 0/0/1
    voice-port 0/2/0
    station-id name POTS
    station-id number 7000
    voice-port 0/2/1
    ccm-manager config
    dial-peer voice 7000 pots
    destination-pattern 5006
    port 0/2/0
    dial-peer voice 90 pots
    description Emergency Services
    destination-pattern 911
    port 0/0/0
    forward-digits 3
    dial-peer voice 91 pots
    description 10 Digit local dialing
    destination-pattern [234].........
    port 0/0/0
    forward-digits 10
    dial-peer voice 92 pots
    description 11 Digit local/long distance dialing
    destination-pattern 1[2348].........
    port 0/0/0
    forward-digits 11
    dial-peer voice 93 pots
    description Long Distance
    destination-pattern 011T
    port 0/0/0
    prefix 011
    dial-peer voice 94 pots
    description Backup bench POTS phone
    destination-pattern 7000
    port 0/2/0
    dial-peer voice 2 voip
    destination-pattern 51..
    session protocol sipv2
    session target ipv4:172.16.2.155
    dtmf-relay sip-notify
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    no vad

    Hi
    Can you find the below:-
    Hi
    1- Please find the below table  as the following link  http://www.cisco.com/en/US/products/hw/routers/ps274/products_tech_note09186a00800b53c7.shtml
    Caller ID          Requires VIC-2FXO-M1, VIC-2FXO-M2, VIC-4FXO-M1, VIC2-2FXO, VIC2-4FXO, or MRP3-8FXOM1
    under voice-port
    caller-id enable
    2-If above configure and still have no caller id , please add the below commannds to the voice-port
    caller-id alerting line-reversal
    cptone ?               "based on your"
    caller-id alerting ring 2    "the default is 1" maximum number of rings to be detected before a call is answered over an FXO voice port.
    4-Do debug to make sure all ok
    "debug vpm signal "
    [0/3/0] get_fxo_caller_id:Caller ID received. Message type=128 length=31 checksum=74
    Thank you
    please rate all useful information

  • 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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