[cisco-voip] Mobility Issue
Nick Matthews
matthnick at gmail.com
Tue Jan 15 19:56:17 EST 2013
It looks like CUCM isn't setting the stream back to active after putting it
on hold. It sends the re-invite, and the 200 OK from the ITSP has the SDP
continued with a=inactive.
I don't have some good traces in front of me, but somewhere it needs to
take that off. I don't think Asterisks is acting incorrectly by responding
to a delayed offer INVITE that was previously a=inactive with a=inactive.
What's also odd is that CUCM is sending the exact same INVITE after the
first one completes, for both the hold and the resume. The CSeq isn't
increasing, which I would expect it to.
If you were to check 'MTP' required it may work around the problem, but I
don't consider inserting MTP's to be a best practice.
-nick
On Tue, Jan 15, 2013 at 3:42 PM, Kenneth Hayes <kennethwhayes at gmail.com>wrote:
> Bind your media and source to your outbound interface on your voice
> service voip.
>
> Sent from my iPhone
>
> On Jan 15, 2013, at 3:39 PM, Dane Newman <dane.newman at gmail.com> wrote:
>
> Below is a show run from the router
>
>
> [OK]
> Cisco3825#sh run
> Building configuration...
>
> Current configuration : 10183 bytes
> !
> ! Last configuration change at 20:49:24 UTC Tue Jan 15 2013 by dnewman
> version 15.1
> service timestamps debug datetime msec
> service timestamps log datetime msec
> no service password-encryption
> !
> hostname Cisco3825
> !
> boot-start-marker
> boot-end-marker
> !
> !
> !
> aaa new-model
> !
> !
> aaa authentication login default local
> aaa authentication login vpnauth local
> aaa authorization exec default local
> aaa authorization network default local
> aaa authorization network vpnauth local
> !
> !
> !
> !
> !
> aaa session-id common
> !
> no network-clock-participate wic 0
> !
> dot11 syslog
> ip source-route
> !
> ip cef
> !
> !
> !
> !
> ip domain name datasc.local
> ip inspect udp idle-time 1800
> no ipv6 cef
> !
> multilink bundle-name authenticated
> !
> !
> !
> !
> !
> voice-card 0
> dsp services dspfarm
> !
> !
> !
> voice service voip
> ip address trusted list
> ipv4 64.154.41.150 255.255.255.255
> allow-connections sip to sip
> fax protocol pass-through g711ulaw
> sip
> !
> !
> !
> !
> voice translation-rule 1
> rule 1 /6784604564/ /200/
> rule 2 /6784563290/ /100/
> rule 3 /6784563291/ /101/
> rule 4 /6784563292/ /102/
> rule 5 /6784563293/ /103/
> rule 6 /6784563294/ /104/
> rule 7 /6784563295/ /105/
> rule 8 /6784563296/ /106/
> rule 9 /6784563297/ /107/
> rule 10 /6784563298/ /108/
> rule 11 /6784563299/ /109/
> rule 12 /6784604565/ /125/
> !
> !
> voice translation-profile incomingdid
> translate called 1
> !
> !
> crypto pki token default removal timeout 0
> !
> crypto pki trustpoint selfsigned
> enrollment selfsigned
> subject-name CN=connect.datasc.com
> revocation-check crl
> !
> !
> crypto pki certificate chain selfsigned
> certificate self-signed 02
> 30820251 308201BA A0030201 02020102 300D0609 2A864886 F70D0101 05050030
> 44311B30 19060355 04031312 636F6E6E 6563742E 64617461 73632E63 6F6D3125
> 30230609 2A864886 F70D0109 02161643 6973636F 33383235 2E646174 6173632E
> 6C6F6361 6C301E17 0D313231 32323831 39313531 395A170D 32303031 30313030
> 30303030 5A304431 1B301906 03550403 1312636F 6E6E6563 742E6461 74617363
> 2E636F6D 31253023 06092A86 4886F70D 01090216 16436973 636F3338 32352E64
> 61746173 632E6C6F 63616C30 819F300D 06092A86 4886F70D 01010105 0003818D
> 00308189 02818100 D9A99B41 8B70C82F 28072967 376E13E8 8F7FC2C2 7729B93E
> DDAE31A0 F3613381 78B43E11 5144BE88 DC2FDE14 0035A104 0BBFAEA0 9A190598
> 19A124B1 2C4A8EA2 04253BA1 C829EF07 CD0E848D E7AA5269 459449C4 FABF9CA9
> BC5AF8ED 84FCD99B 260C2B75 57887863 7BB310FB 2C8D1506 FE91FEAC 4EDD1712
> A7AFD2C1 BF21C59D 02030100 01A35330 51300F06 03551D13 0101FF04 05300301
> 01FF301F 0603551D 23041830 16801475 02C4FB04 4FB3F748 B4012EC5 8A571236
> A190CB30 1D060355 1D0E0416 04147502 C4FB044F B3F748B4 012EC58A 571236A1
> 90CB300D 06092A86 4886F70D 01010505 00038181 00C2B167 E583F6D8 8B742D4F
> 49D27AAD 7EF4E64F 0B5CA5A3 944E8CC7 499A706F AB22283B AE5913A1 B22BBB20
> E7CF6F9F 41CDD870 1B474E58 9537C1FA 2D93BE4F 4276E9CE 61AE18D3 EF724BD9
> 33878860 4B3627C0 448C652D 03D4C142 BA3291A3 DDE0C4DD C6ED06C3 12E45933
> F1EE5CC2 B5B6CC20 C65AB313 76966F14 AA25CC8D 2A
> quit
> !
> !
> license udi pid CISCO3825 sn FTX1237A1T0
> username xxxxxxx privilege 15 secret xxxxxx
> !
> redundancy
> !
> !
> controller T1 0/0/0
> !
> controller T1 0/0/1
> !
> ip ssh version 2
> !
> !
> crypto isakmp policy 10
> encr aes
> authentication pre-share
> group 2
> crypto isakmp key Recoil90 address 0.0.0.0 0.0.0.0
> crypto isakmp fragmentation
> !
> crypto isakmp client configuration group datasc
> key Recoil90
> dns 4.2.2.2 4.2.2.1
> domain datasc.local
> pool vpnpool
> save-password
> !
> crypto isakmp client configuration group datascsplit
> key Recoil90
> dns 4.2.2.2 4.2.2.1
> domain datasc.local
> pool vpnpool
> acl 101
> save-password
> crypto isakmp profile datasc
> match identity group datasc
> client authentication list vpnauth
> isakmp authorization list vpnauth
> client configuration address respond
> virtual-template 1
> crypto isakmp profile datascsplit
> match identity group datascsplit
> client authentication list vpnauth
> isakmp authorization list vpnauth
> client configuration address respond
> virtual-template 2
> !
> !
> crypto ipsec transform-set transformset esp-aes
> crypto ipsec transform-set ezvpntransform esp-aes esp-sha-hmac
> !
> crypto ipsec profile VTI
> set transform-set ezvpntransform
> set isakmp-profile datasc
> !
> crypto ipsec profile VTI2
> set transform-set ezvpntransform
> set isakmp-profile datascsplit
> !
> !
> !
> !
> !
> !
> !
> interface Loopback1
> ip address 10.1.150.1 255.255.255.0
> ip nat inside
> ip virtual-reassembly in
> !
> interface GigabitEthernet0/0
> ip address dhcp
> no ip redirects
> no ip unreachables
> no ip proxy-arp
> ip nat outside
> ip virtual-reassembly in
> duplex auto
> speed auto
> media-type rj45
> hold-queue 240000 in
> !
> interface GigabitEthernet0/1
> ip address 10.1.200.1 255.255.255.252
> ip nat inside
> ip virtual-reassembly in
> duplex auto
> speed auto
> media-type rj45
> !
> interface Virtual-Template1 type tunnel
> ip unnumbered GigabitEthernet0/0
> ip nat inside
> ip virtual-reassembly in
> tunnel source GigabitEthernet0/0
> tunnel mode ipsec ipv4
> tunnel protection ipsec profile VTI
> !
> interface Virtual-Template2 type tunnel
> ip unnumbered GigabitEthernet0/0
> ip nat inside
> ip virtual-reassembly in
> tunnel source GigabitEthernet0/0
> tunnel mode ipsec ipv4
> tunnel protection ipsec profile VTI2
> !
> interface Virtual-Template3
> ip unnumbered GigabitEthernet0/0
> ip nat outside
> ip virtual-reassembly in
> ip policy route-map anyconnecthop
> !
> !
> router eigrp 1
> maximum-paths 1
> network 10.0.0.0
> redistribute static
> !
> ip local pool vpnpool 10.1.100.10 10.1.100.200
> ip forward-protocol nd
> ip http server
> ip http secure-server
> !
> !
> ip nat inside source list NATNETWORKS interface GigabitEthernet0/0 overload
> ip nat inside source static tcp 10.1.50.150 80 interface
> GigabitEthernet0/0 80
> ip nat inside source static tcp 10.1.80.100 5001 interface
> GigabitEthernet0/0 5001
> ip nat inside source static udp 10.1.80.100 5001 interface
> GigabitEthernet0/0 5001
> !
> ip access-list extended NATNETWORKS
> deny ip 10.0.0.0 0.255.255.255 172.16.0.0 0.15.255.255
> deny ip 10.0.0.0 0.255.255.255 10.0.0.0 0.255.255.255
> permit ip 10.0.0.0 0.255.255.255 any
> ip access-list extended anyconnecthop
> deny ip 10.0.0.0 0.255.255.255 10.0.0.0 0.255.255.255
> permit ip 10.0.0.0 0.255.255.255 any
> !
> access-list 50 permit 10.0.0.0 0.255.255.255
> access-list 101 permit ip 10.0.0.0 0.255.255.255 any
> !
> !
> !
> !
> route-map anyconnecthop permit 10
> match ip address anyconnecthop
> set ip next-hop 10.1.150.2
> !
> !
> !
> !
> !
> control-plane
> !
> !
> !
> !
> mgcp profile default
> !
> !
> dial-peer voice 100 voip
> description Publisher
> preference 1
> destination-pattern 1..
> session protocol sipv2
> session target ipv4:10.1.80.10
> dtmf-relay rtp-nte
> codec g711ulaw
> !
> dial-peer voice 101 voip
> description Subscriber
> preference 2
> destination-pattern 1..
> session target ipv4:10.1.80.11
> dtmf-relay rtp-nte
> codec g711ulaw
> !
> dial-peer voice 200 voip
> description Publisher
> preference 1
> destination-pattern 2..
> progress_ind setup enable 3
> progress_ind progress enable 8
> session protocol sipv2
> session target ipv4:10.1.80.10
> dtmf-relay rtp-nte
> codec g711ulaw
> !
> dial-peer voice 300 voip
> description incoming Calldid
> translation-profile incoming incomingdid
> preference 1
> session protocol sipv2
> session target sip-server
> incoming called-number 678456329.
> dtmf-relay rtp-nte
> codec g711ulaw
> !
> dial-peer voice 301 voip
> description incoming Calldid
> translation-profile incoming incomingdid
> preference 1
> session protocol sipv2
> session target sip-server
> incoming called-number 6784604565
> dtmf-relay rtp-nte
> codec g711ulaw
> !
> dial-peer voice 302 voip
> description incoming Calldid
> translation-profile incoming incomingdid
> preference 1
> session protocol sipv2
> session target sip-server
> incoming called-number 6784604564
> dtmf-relay rtp-nte
> codec g711ulaw
> !
> dial-peer voice 201 voip
> description Publisher
> preference 2
> destination-pattern 2..
> progress_ind setup enable 3
> progress_ind progress enable 8
> session protocol sipv2
> session target ipv4:10.1.80.11
> dtmf-relay rtp-nte
> codec g711ulaw
> !
> dial-peer voice 500 voip
> description outgoing
> preference 1
> destination-pattern .T
> session protocol sipv2
> session target dns:sip.talkinip.net
> dtmf-relay rtp-nte
> codec g711ulaw
> !
> !
> sip-ua
> credentials username xxxxxxxx password 7 xxxxxxx realm
> sipconnect.ipcomms.net
> authentication username xxxxxx password 7 xxxxxxx
> authentication username xxxxx password 7 xxxxxxx realm
> sipconnect.ipcomms.net
> set pstn-cause 3 sip-status 486
> set pstn-cause 34 sip-status 486
> set pstn-cause 47 sip-status 486
> registrar dns:sipconnect.ipcomms.net expires 60
> sip-server dns:sipconnect.ipcomms.net:5060
> !
> !
> !
> gatekeeper
> shutdown
> !
> !
> call-manager-fallback
> max-conferences 8 gain -6
> transfer-system full-consult
> ip source-address 10.1.200.1 port 2000
> max-ephones 20
> max-dn 40
> !
> !
> !
> line con 0
> line aux 0
> line vty 0 4
> privilege level 15
> transport input ssh
> line vty 5 15
> privilege level 15
> transport input ssh
> !
> scheduler allocate 20000 1000
> !
> webvpn gateway gateway_1
> ip interface GigabitEthernet0/0 port 443
> ssl trustpoint selfsigned
> inservice
> !
> webvpn install svc flash:/webvpn/anyconnect-win-3.1.02026-k9.pkg sequence
> 1
> !
> webvpn context datasc
> title "DataSC VPN"
> secondary-color white
> title-color #CCCC66
> text-color black
> ssl authenticate verify all
> !
> url-list "Servers"
> heading "Server"
> !
> !
> policy group datasc
> url-list "Servers"
> functions svc-enabled
> svc address-pool "vpnpool" netmask 255.255.255.0
> svc keep-client-installed
> svc dns-server primary 4.2.2.2
> svc dtls
> virtual-template 3
> default-group-policy datasc
> aaa authentication list vpnauth
> gateway gateway_1 domain datasc
> inservice
> !
> !
> webvpn context datascsplit
> title "DataSC VPN Split"
> secondary-color white
> title-color #CCCC66
> text-color black
> ssl authenticate verify all
> !
> url-list "Servers"
> heading "Server"
> !
> !
> policy group datascsplit
> url-list "Servers"
> functions svc-enabled
> svc address-pool "vpnpool" netmask 255.255.255.0
> svc split include acl 50
> svc dns-server primary 4.2.2.2
> svc dtls
> default-group-policy datascsplit
> aaa authentication list vpnauth
> gateway gateway_1 domain datascsplit
> inservice
> !
> end
> Cisco3825#
>
> On Tue, Jan 15, 2013 at 3:31 PM, Kenneth Hayes <kennethwhayes at gmail.com>wrote:
>
>> What do your media resources look like?
>>
>>
>> Also can you show me a copy of your voice service voip config?
>>
>> Sent from my iPad
>>
>> On Jan 15, 2013, at 3:12 PM, Dane Newman <dane.newman at gmail.com> wrote:
>>
>> Thanks Ryan
>>
>> I see I am always getting a 200 ok message after my invites from the debug
>>
>> *Putting a call on HOLD*
>>
>>
>>
>> *Jan 15 20:19:28.086: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>
>> Received:
>>
>> INVITE sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0
>>
>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK2891b89f8fa
>>
>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>
>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>
>> Date: Tue, 15 Jan 2013 19:57:35 GMT
>>
>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>
>> Supported: timer,resource-priority,replaces
>>
>> Min-SE: 1800
>>
>> User-Agent: Cisco-CUCM8.6
>>
>> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>> SUBSCRIBE, NOTIFY
>>
>> CSeq: 102 INVITE
>>
>> Max-Forwards: 70
>>
>> Expires: 180
>>
>> Allow-Events: presence
>>
>> Supported: X-cisco-srtp-fallback
>>
>> Supported: Geolocation
>>
>> Session-Expires: 1800;refresher=uas
>>
>> P-Asserted-Identity: "Dane Newman" <sip:6784563290 at 10.1.80.10>
>>
>> Remote-Party-ID: "Dane Newman" <sip:6784563290 at 10.1.80.10
>> >;party=calling;screen=yes;privacy=off
>>
>> Contact: <sip:6784563290 at 10.1.80.10:5060;transport=tcp>;video
>>
>> Content-Type: application/sdp
>>
>> Content-Length: 240
>>
>> v=0
>>
>> o=CiscoSystemsCCM-SIP 7322 3 IN IP4 10.1.80.10
>>
>> s=SIP Call
>>
>> c=IN IP4 0.0.0.0
>>
>> b=TIAS:64000
>>
>> b=AS:64
>>
>> t=0 0
>>
>> m=audio 21476 RTP/AVP 0 101
>>
>> a=rtpmap:0 PCMU/8000
>>
>> a=ptime:20
>>
>> a=inactive
>>
>> a=rtpmap:101 telephone-event/8000
>>
>> a=fmtp:101 0-15
>>
>> *Jan 15 20:19:28.094: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>
>> Sent:
>>
>> INVITE sip:17705439047 at 64.154.41.150:5060 SIP/2.0
>>
>> Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK691F12E0
>>
>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>> >;tag=2E6BC0B0-2268
>>
>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>
>> Date: Tue, 15 Jan 2013 20:19:28 GMT
>>
>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>
>> Supported: 100rel,timer,resource-priority,replaces,sdp-anat
>>
>> Min-SE: 1800
>>
>> Cisco-Guid: 3257897472-0000065536-0000000035-0173015306
>>
>> User-Agent: Cisco-SIPGateway/IOS-12.x
>>
>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>>
>> CSeq: 103 INVITE
>>
>> Max-Forwards: 70
>>
>> Timestamp: 1358281168
>>
>> Contact: <sip:6784563290 at 98.192.104.214:5060>
>>
>> Expires: 180
>>
>> Allow-Events: telephone-event
>>
>> Authorization: Digest username="6784563290",realm="asterisk",uri="
>> sip:17705439047 at 64.154.41.150:5060
>> ",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5
>>
>> Session-Expires: 1800;refresher=uas
>>
>> Content-Type: application/sdp
>>
>> Content-Length: 289
>>
>> v=0
>>
>> o=CiscoSystemsSIP-GW-UserAgent 3168 2739 IN IP4 98.192.104.214
>>
>> s=SIP Call
>>
>> c=IN IP4 98.192.104.214
>>
>> t=0 0
>>
>> m=audio 19458 RTP/AVP 0 101 19
>>
>> c=IN IP4 98.192.104.214
>>
>> a=inactive
>>
>> a=rtpmap:0 PCMU/8000
>>
>> a=rtpmap:101 telephone-event/8000
>>
>> a=fmtp:101 0-15
>>
>> a=rtpmap:19 CN/8000
>>
>> a=ptime:20
>>
>> *Jan 15 20:19:28.094: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>
>> Sent:
>>
>> SIP/2.0 100 Trying
>>
>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK2891b89f8fa
>>
>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>
>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>
>> Date: Tue, 15 Jan 2013 20:19:28 GMT
>>
>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>
>> CSeq: 102 INVITE
>>
>> Allow-Events: telephone-event
>>
>> Server: Cisco-SIPGateway/IOS-12.x
>>
>> Content-Length: 0
>>
>> *Jan 15 20:19:28.110: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>
>> Received:
>>
>> SIP/2.0 100 Trying
>>
>> Via: SIP/2.0/UDP 98.192.104.214:5060
>> ;branch=z9hG4bK691F12E0;received=98.192.104.214
>>
>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>> >;tag=2E6BC0B0-2268
>>
>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>
>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>
>> CSeq: 103 INVITE
>>
>> Server: Asterisk PBX 1.6.2.13
>>
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
>>
>> Supported: replaces, timer
>>
>> Require: timer
>>
>> Session-Expires: 1800;refresher=uas
>>
>> Contact: <sip:17705439047 at 64.154.41.150>
>>
>> Content-Length: 0
>>
>> *Jan 15 20:19:28.110: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>
>> Received:
>>
>> SIP/2.0 200 OK
>>
>> Via: SIP/2.0/UDP 98.192.104.214:5060
>> ;branch=z9hG4bK691F12E0;received=98.192.104.214
>>
>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>> >;tag=2E6BC0B0-2268
>>
>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>
>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>
>> CSeq: 103 INVITE
>>
>> Server: Asterisk PBX 1.6.2.13
>>
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
>>
>> Supported: replaces, timer
>>
>> Require: timer
>>
>> Session-Expires: 1800;refresher=uas
>>
>> Contact: <sip:17705439047 at 64.154.41.150>
>>
>> Content-Type: application/sdp
>>
>> Content-Length: 239
>>
>> v=0
>>
>> o=root 1685873050 1685873052 IN IP4 64.154.41.150
>>
>> s=Asterisk PBX 1.6.2.13
>>
>> c=IN IP4 64.154.41.150
>>
>> t=0 0
>>
>> m=audio 13014 RTP/AVP 0 101
>>
>> a=rtpmap:0 PCMU/8000
>>
>> a=rtpmap:101 telephone-event/8000
>>
>> a=fmtp:101 0-16
>>
>> a=ptime:20
>>
>> a=inactive
>>
>> *Jan 15 20:19:28.118: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>
>> Sent:
>>
>> SIP/2.0 200 OK
>>
>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK2891b89f8fa
>>
>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>
>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>
>> Date: Tue, 15 Jan 2013 20:19:28 GMT
>>
>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>
>> CSeq: 102 INVITE
>>
>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>>
>> Allow-Events: telephone-event
>>
>> Remote-Party-ID: <sip:17705439047 at 10.1.200.1
>> >;party=called;screen=no;privacy=off
>>
>> Contact: <sip:17705439047 at 10.1.200.1:5060;transport=tcp>
>>
>> Supported: replaces
>>
>> Supported: sdp-anat
>>
>> Server: Cisco-SIPGateway/IOS-12.x
>>
>> Session-Expires: 1800;refresher=uas
>>
>> Require: timer
>>
>> Supported: timer
>>
>> Content-Type: application/sdp
>>
>> Content-Length: 253
>>
>> v=0
>>
>> o=CiscoSystemsSIP-GW-UserAgent 4444 5479 IN IP4 10.1.200.1
>>
>> s=SIP Call
>>
>> c=IN IP4 10.1.200.1
>>
>> t=0 0
>>
>> m=audio 19514 RTP/AVP 0 101
>>
>> c=IN IP4 10.1.200.1
>>
>> a=inactive
>>
>> a=rtpmap:0 PCMU/8000
>>
>> a=rtpmap:101 telephone-event/8000
>>
>> a=fmtp:101 0-16
>>
>> a=ptime:20
>>
>> *Jan 15 20:19:28.118: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>
>> Sent:
>>
>> ACK sip:17705439047 at 64.154.41.150:5060 SIP/2.0
>>
>> Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK6920266D
>>
>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>> >;tag=2E6BC0B0-2268
>>
>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>
>> Date: Tue, 15 Jan 2013 20:19:28 GMT
>>
>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>
>> Max-Forwards: 70
>>
>> CSeq: 103 ACK
>>
>> Authorization: Digest username="6784563290",realm="asterisk",uri="
>> sip:17705439047 at 64.154.41.150:5060
>> ",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5
>>
>> Allow-Events: telephone-event
>>
>> Content-Length: 0
>>
>> *Jan 15 20:19:28.122: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>
>> Received:
>>
>> ACK sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0
>>
>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28b4b1305a0
>>
>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>
>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>
>> Date: Tue, 15 Jan 2013 19:57:35 GMT
>>
>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>
>> Max-Forwards: 70
>>
>> CSeq: 102 ACK
>>
>> Allow-Events: presence
>>
>> Content-Length: 0
>>
>> *Jan 15 20:19:28.122: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>
>> Received:
>>
>> INVITE sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0
>>
>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28c43b47e7e
>>
>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>
>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>
>> Date: Tue, 15 Jan 2013 19:57:35 GMT
>>
>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>
>> Supported: timer,resource-priority,replaces
>>
>> Min-SE: 1800
>>
>> User-Agent: Cisco-CUCM8.6
>>
>> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>> SUBSCRIBE, NOTIFY
>>
>> CSeq: 103 INVITE
>>
>> Max-Forwards: 70
>>
>> Expires: 180
>>
>> Allow-Events: presence
>>
>> Supported: X-cisco-srtp-fallback
>>
>> Supported: Geolocation
>>
>> Session-Expires: 1800;refresher=uas
>>
>> P-Asserted-Identity: "Dane Newman" <sip:6784563290 at 10.1.80.10>
>>
>> Remote-Party-ID: "Dane Newman" <sip:6784563290 at 10.1.80.10
>> >;party=calling;screen=yes;privacy=off
>>
>> Contact: <sip:6784563290 at 10.1.80.10:5060;transport=tcp>;video
>>
>> Content-Length: 0
>>
>> *Jan 15 20:19:28.126: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>
>> Sent:
>>
>> INVITE sip:17705439047 at 64.154.41.150:5060 SIP/2.0
>>
>> Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69211AB3
>>
>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>> >;tag=2E6BC0B0-2268
>>
>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>
>> Date: Tue, 15 Jan 2013 20:19:28 GMT
>>
>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>
>> Supported: timer,resource-priority,replaces,sdp-anat
>>
>> Min-SE: 1800
>>
>> Cisco-Guid: 3257897472-0000065536-0000000035-0173015306
>>
>> User-Agent: Cisco-SIPGateway/IOS-12.x
>>
>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>>
>> CSeq: 104 INVITE
>>
>> Max-Forwards: 70
>>
>> Timestamp: 1358281168
>>
>> Contact: <sip:6784563290 at 98.192.104.214:5060>
>>
>> Expires: 180
>>
>> Allow-Events: telephone-event
>>
>> Authorization: Digest username="6784563290",realm="asterisk",uri="
>> sip:17705439047 at 64.154.41.150:5060
>> ",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5
>>
>> Session-Expires: 1800;refresher=uas
>>
>> Content-Length: 0
>>
>> *Jan 15 20:19:28.126: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>
>> Sent:
>>
>> SIP/2.0 100 Trying
>>
>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28c43b47e7e
>>
>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>
>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>
>> Date: Tue, 15 Jan 2013 20:19:28 GMT
>>
>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>
>> CSeq: 103 INVITE
>>
>> Allow-Events: telephone-event
>>
>> Server: Cisco-SIPGateway/IOS-12.x
>>
>> Content-Length: 0
>>
>> *Jan 15 20:19:28.146: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>
>> Received:
>>
>> SIP/2.0 100 Trying
>>
>> Via: SIP/2.0/UDP 98.192.104.214:5060
>> ;branch=z9hG4bK69211AB3;received=98.192.104.214
>>
>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>> >;tag=2E6BC0B0-2268
>>
>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>
>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>
>> CSeq: 104 INVITE
>>
>> Server: Asterisk PBX 1.6.2.13
>>
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
>>
>> Supported: replaces, timer
>>
>> Require: timer
>>
>> Session-Expires: 1800;refresher=uas
>>
>> Contact: <sip:17705439047 at 64.154.41.150>
>>
>> Content-Length: 0
>>
>> *Jan 15 20:19:28.146: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>
>> Received:
>>
>> SIP/2.0 200 OK
>>
>> Via: SIP/2.0/UDP 98.192.104.214:5060
>> ;branch=z9hG4bK69211AB3;received=98.192.104.214
>>
>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>> >;tag=2E6BC0B0-2268
>>
>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>
>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>
>> CSeq: 104 INVITE
>>
>> Server: Asterisk PBX 1.6.2.13
>>
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
>>
>> Supported: replaces, timer
>>
>> Require: timer
>>
>> Session-Expires: 1800;refresher=uas
>>
>> Contact: <sip:17705439047 at 64.154.41.150>
>>
>> Content-Type: application/sdp
>>
>> Content-Length: 333
>>
>> v=0
>>
>> o=root 1685873050 1685873053 IN IP4 64.154.41.150
>>
>> s=Asterisk PBX 1.6.2.13
>>
>> c=IN IP4 64.154.41.150
>>
>> t=0 0
>>
>> m=audio 13014 RTP/AVP 3 8 0 18 101
>>
>> a=rtpmap:3 GSM/8000
>>
>> a=rtpmap:8 PCMA/8000
>>
>> a=rtpmap:0 PCMU/8000
>>
>> a=rtpmap:18 G729/8000
>>
>> a=fmtp:18 annexb=no
>>
>> a=rtpmap:101 telephone-event/8000
>>
>> a=fmtp:101 0-16
>>
>> a=ptime:20
>>
>> a=inactive
>>
>> *Jan 15 20:19:28.150: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>
>> Sent:
>>
>> SIP/2.0 200 OK
>>
>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28c43b47e7e
>>
>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>
>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>
>> Date: Tue, 15 Jan 2013 20:19:28 GMT
>>
>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>
>> CSeq: 103 INVITE
>>
>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>>
>> Allow-Events: telephone-event
>>
>> Remote-Party-ID: <sip:17705439047 at 10.1.200.1
>> >;party=called;screen=no;privacy=off
>>
>> Contact: <sip:17705439047 at 10.1.200.1:5060;transport=tcp>
>>
>> Supported: replaces
>>
>> Supported: sdp-anat
>>
>> Server: Cisco-SIPGateway/IOS-12.x
>>
>> Session-Expires: 1800;refresher=uas
>>
>> Require: timer
>>
>> Supported: timer
>>
>> Content-Type: application/sdp
>>
>> Content-Length: 277
>>
>> v=0
>>
>> o=CiscoSystemsSIP-GW-UserAgent 4444 5480 IN IP4 10.1.200.1
>>
>> s=SIP Call
>>
>> c=IN IP4 10.1.200.1
>>
>> t=0 0
>>
>> m=audio 19514 RTP/AVP 0 101 19
>>
>> c=IN IP4 10.1.200.1
>>
>> a=inactive
>>
>> a=rtpmap:0 PCMU/8000
>>
>> a=rtpmap:101 telephone-event/8000
>>
>> a=fmtp:101 0-16
>>
>> a=rtpmap:19 CN/8000
>>
>> a=ptime:20
>>
>> *Jan 15 20:19:28.162: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>
>> Received:
>>
>> ACK sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0
>>
>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28d3eadaab3
>>
>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>
>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>
>> Date: Tue, 15 Jan 2013 19:57:35 GMT
>>
>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>
>> Max-Forwards: 70
>>
>> CSeq: 103 ACK
>>
>> Allow-Events: presence
>>
>> Content-Type: application/sdp
>>
>> Content-Length: 209
>>
>> v=0
>>
>> o=CiscoSystemsCCM-SIP 7322 4 IN IP4 10.1.80.10
>>
>> s=SIP Call
>>
>> c=IN IP4 0.0.0.0
>>
>> b=TIAS:64000
>>
>> b=AS:64
>>
>> t=0 0
>>
>> m=audio 21476 RTP/AVP 0
>>
>> a=X-cisco-media:nomedia
>>
>> a=rtpmap:0 PCMU/8000
>>
>> a=ptime:20
>>
>> a=inactive
>>
>> *Jan 15 20:19:28.170: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>
>> Sent:
>>
>> ACK sip:17705439047 at 64.154.41.150:5060 SIP/2.0
>>
>> Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK692226EA
>>
>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>> >;tag=2E6BC0B0-2268
>>
>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>
>> Date: Tue, 15 Jan 2013 20:19:28 GMT
>>
>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>
>> Max-Forwards: 70
>>
>> CSeq: 104 ACK
>>
>> Authorization: Digest username="6784563290",realm="asterisk",uri="
>> sip:17705439047 at 64.154.41.150:5060
>> ",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5
>>
>> Allow-Events: telephone-event
>>
>> Content-Type: application/sdp
>>
>> Content-Length: 251
>>
>> v=0
>>
>> o=CiscoSystemsSIP-GW-UserAgent 3168 2740 IN IP4 98.192.104.214
>>
>> s=SIP Call
>>
>> c=IN IP4 0.0.0.0
>>
>> t=0 0
>>
>> m=audio 19458 RTP/AVP 0 101
>>
>> c=IN IP4 0.0.0.0
>>
>> a=inactive
>>
>> a=rtpmap:0 PCMU/8000
>>
>> a=rtpmap:101 telephone-event/8000
>>
>> a=fmtp:101 0-16
>>
>> a=ptime:20
>>
>>
>>
>> *Unholding the call the MOH continues on the previously held caller
>> while the user hears nothing*
>>
>> **
>>
>>
>>
>> *Jan 15 20:19:35.166: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>
>> Received:
>>
>> INVITE sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0
>>
>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c
>>
>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>
>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>
>> Date: Tue, 15 Jan 2013 19:57:42 GMT
>>
>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>
>> Supported: timer,resource-priority,replaces
>>
>> Min-SE: 1800
>>
>> User-Agent: Cisco-CUCM8.6
>>
>> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>> SUBSCRIBE, NOTIFY
>>
>> CSeq: 104 INVITE
>>
>> Max-Forwards: 70
>>
>> Expires: 180
>>
>> Allow-Events: presence
>>
>> Supported: X-cisco-srtp-fallback
>>
>> Supported: Geolocation
>>
>> Session-Expires: 1800;refresher=uas
>>
>> P-Asserted-Identity: "Dane Newman" <sip:6784563290 at 10.1.80.10>
>>
>> Remote-Party-ID: "Dane Newman" <sip:6784563290 at 10.1.80.10
>> >;party=calling;screen=yes;privacy=off
>>
>> Contact: <sip:6784563290 at 10.1.80.10:5060;transport=tcp>;video;audio;video
>>
>> Content-Length: 0
>>
>> *Jan 15 20:19:35.170: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>
>> Sent:
>>
>> INVITE sip:17705439047 at 64.154.41.150:5060 SIP/2.0
>>
>> Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69232672
>>
>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>> >;tag=2E6BC0B0-2268
>>
>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>
>> Date: Tue, 15 Jan 2013 20:19:35 GMT
>>
>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>
>> Supported: timer,resource-priority,replaces,sdp-anat
>>
>> Min-SE: 1800
>>
>> Cisco-Guid: 3257897472-0000065536-0000000035-0173015306
>>
>> User-Agent: Cisco-SIPGateway/IOS-12.x
>>
>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>>
>> CSeq: 105 INVITE
>>
>> Max-Forwards: 70
>>
>> Timestamp: 1358281175
>>
>> Contact: <sip:6784563290 at 98.192.104.214:5060>
>>
>> Expires: 180
>>
>> Allow-Events: telephone-event
>>
>> Authorization: Digest username="6784563290",realm="asterisk",uri="
>> sip:17705439047 at 64.154.41.150:5060
>> ",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5
>>
>> Session-Expires: 1800;refresher=uas
>>
>> Content-Length: 0
>>
>> *Jan 15 20:19:35.190: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>
>> Sent:
>>
>> SIP/2.0 100 Trying
>>
>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c
>>
>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>
>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>
>> Date: Tue, 15 Jan 2013 20:19:35 GMT
>>
>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>
>> CSeq: 104 INVITE
>>
>> Allow-Events: telephone-event
>>
>> Server: Cisco-SIPGateway/IOS-12.x
>>
>> Content-Length: 0
>>
>> *Jan 15 20:19:35.194: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>
>> Received:
>>
>> SIP/2.0 100 Trying
>>
>> Via: SIP/2.0/UDP 98.192.104.214:5060
>> ;branch=z9hG4bK69232672;received=98.192.104.214
>>
>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>> >;tag=2E6BC0B0-2268
>>
>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>
>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>
>> CSeq: 105 INVITE
>>
>> Server: Asterisk PBX 1.6.2.13
>>
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
>>
>> Supported: replaces, timer
>>
>> Require: timer
>>
>> Session-Expires: 1800;refresher=uas
>>
>> Contact: <sip:17705439047 at 64.154.41.150>
>>
>> Content-Length: 0
>>
>> *Jan 15 20:19:35.194: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>
>> Received:
>>
>> SIP/2.0 200 OK
>>
>> Via: SIP/2.0/UDP 98.192.104.214:5060
>> ;branch=z9hG4bK69232672;received=98.192.104.214
>>
>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>> >;tag=2E6BC0B0-2268
>>
>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>
>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>
>> CSeq: 105 INVITE
>>
>> Server: Asterisk PBX 1.6.2.13
>>
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
>>
>> Supported: replaces, timer
>>
>> Require: timer
>>
>> Session-Expires: 1800;refresher=uas
>>
>> Contact: <sip:17705439047 at 64.154.41.150>
>>
>> Content-Type: application/sdp
>>
>> Content-Length: 333
>>
>> v=0
>>
>> o=root 1685873050 1685873054 IN IP4 64.154.41.150
>>
>> s=Asterisk PBX 1.6.2.13
>>
>> c=IN IP4 64.154.41.150
>>
>> t=0 0
>>
>> m=audio 13014 RTP/AVP 3 8 0 18 101
>>
>> a=rtpmap:3 GSM/8000
>>
>> a=rtpmap:8 PCMA/8000
>>
>> a=rtpmap:0 PCMU/8000
>>
>> a=rtpmap:18 G729/8000
>>
>> a=fmtp:18 annexb=no
>>
>> a=rtpmap:101 telephone-event/8000
>>
>> a=fmtp:101 0-16
>>
>> a=ptime:20
>>
>> a=inactive
>>
>> *Jan 15 20:19:35.198: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>
>> Sent:
>>
>> SIP/2.0 200 OK
>>
>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c
>>
>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>
>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>
>> Date: Tue, 15 Jan 2013 20:19:35 GMT
>>
>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>
>> CSeq: 104 INVITE
>>
>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>>
>> Allow-Events: telephone-event
>>
>> Remote-Party-ID: <sip:17705439047 at 10.1.200.1
>> >;party=called;screen=no;privacy=off
>>
>> Contact: <sip:17705439047 at 10.1.200.1:5060;transport=tcp>
>>
>> Supported: replaces
>>
>> Supported: sdp-anat
>>
>> Server: Cisco-SIPGateway/IOS-12.x
>>
>> Session-Expires: 1800;refresher=uas
>>
>> Require: timer
>>
>> Supported: timer
>>
>> Content-Type: application/sdp
>>
>> Content-Length: 277
>>
>> v=0
>>
>> o=CiscoSystemsSIP-GW-UserAgent 4444 5481 IN IP4 10.1.200.1
>>
>> s=SIP Call
>>
>> c=IN IP4 10.1.200.1
>>
>> t=0 0
>>
>> m=audio 19514 RTP/AVP 0 101 19
>>
>> c=IN IP4 10.1.200.1
>>
>> a=inactive
>>
>> a=rtpmap:0 PCMU/8000
>>
>> a=rtpmap:101 telephone-event/8000
>>
>> a=fmtp:101 0-16
>>
>> a=rtpmap:19 CN/8000
>>
>> a=ptime:20
>>
>> *Jan 15 20:19:35.206: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>
>> Received:
>>
>> ACK sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0
>>
>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28f6dca6616
>>
>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>
>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>
>> Date: Tue, 15 Jan 2013 19:57:42 GMT
>>
>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>
>> Max-Forwards: 70
>>
>> CSeq: 104 ACK
>>
>> Allow-Events: presence, kpml
>>
>> Content-Type: application/sdp
>>
>> Content-Length: 243
>>
>> v=0
>>
>> o=CiscoSystemsCCM-SIP 7322 5 IN IP4 10.1.80.10
>>
>> s=SIP Call
>>
>> c=IN IP4 10.1.10.18
>>
>> b=TIAS:64000
>>
>> b=AS:64
>>
>> t=0 0
>>
>> m=audio 21476 RTP/AVP 0 101
>>
>> a=rtpmap:0 PCMU/8000
>>
>> a=ptime:20
>>
>> a=inactive
>>
>> a=rtpmap:101 telephone-event/8000
>>
>> a=fmtp:101 0-15
>>
>> *Jan 15 20:19:35.210: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>
>> Sent:
>>
>> ACK sip:17705439047 at 64.154.41.150:5060 SIP/2.0
>>
>> Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69246AB
>>
>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>> >;tag=2E6BC0B0-2268
>>
>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>
>> Date: Tue, 15 Jan 2013 20:19:35 GMT
>>
>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>
>> Max-Forwards: 70
>>
>> CSeq: 105 ACK
>>
>> Authorization: Digest username="6784563290",realm="asterisk",uri="
>> sip:17705439047 at 64.154.41.150:5060
>> ",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5
>>
>> Allow-Events: telephone-event
>>
>> Content-Type: application/sdp
>>
>> Content-Length: 265
>>
>> v=0
>>
>> o=CiscoSystemsSIP-GW-UserAgent 3168 2741 IN IP4 98.192.104.214
>>
>> s=SIP Call
>>
>> c=IN IP4 98.192.104.214
>>
>> t=0 0
>>
>> m=audio 19458 RTP/AVP 0 101
>>
>> c=IN IP4 98.192.104.214
>>
>> a=inactive
>>
>> a=rtpmap:0 PCMU/8000
>>
>> a=rtpmap:101 telephone-event/8000
>>
>> a=fmtp:101 0-16
>>
>> a=ptime:20
>>
>> Cisco3825#
>>
>> Cisco3825#
>>
>>
>>
>> Cisco3825#
>>
>>
>>
>> INVITE sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0
>>
>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c
>>
>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>
>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>
>> Date: Tue, 15 Jan 2013 19:57:42 GMT
>>
>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>
>> Supported: timer,resource-priority,replaces
>>
>> Min-SE: 1800
>>
>> User-Agent: Cisco-CUCM8.6
>>
>> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>> SUBSCRIBE, NOTIFY
>>
>> CSeq: 104 INVITE
>>
>> Max-Forwards: 70
>>
>> Expires: 180
>>
>> Allow-Events: presence
>>
>> Supported: X-cisco-srtp-fallback
>>
>> Supported: Geolocation
>>
>> Session-Expires: 1800;refresher=uas
>>
>> P-Asserted-Identity: "Dane Newman" <sip:6784563290 at 10.1.80.10>
>>
>> Remote-Party-ID: "Dane Newman" <sip:6784563290 at 10.1.80.10
>> >;party=calling;screen=yes;privacy=off
>>
>> Contact: <sip:6784563290 at 10.1.80.10:5060;transport=tcp>;video;audio;video
>>
>> Content-Length: 0
>>
>> *Jan 15 20:19:35.170: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>
>> Sent:
>>
>> INVITE sip:17705439047 at 64.154.41.150:5060 SIP/2.0
>>
>> Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69232672
>>
>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>> >;tag=2E6BC0B0-2268
>>
>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>
>> Date: Tue, 15 Jan 2013 20:19:35 GMT
>>
>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>
>> Supported: timer,resource-priority,replaces,sdp-anat
>>
>> Min-SE: 1800
>>
>> Cisco-Guid: 3257897472-0000065536-0000000035-0173015306
>>
>> User-Agent: Cisco-SIPGateway/IOS-12.x
>>
>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>>
>> CSeq: 105 INVITE
>>
>> Max-Forwards: 70
>>
>> Timestamp: 1358281175
>>
>> Contact: <sip:6784563290 at 98.192.104.214:5060>
>>
>> Expires: 180
>>
>> Allow-Events: telephone-event
>>
>> Authorization: Digest username="6784563290",realm="asterisk",uri="
>> sip:17705439047 at 64.154.41.150:5060
>> ",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5
>>
>> Session-Expires: 1800;refresher=uas
>>
>> Content-Length: 0
>>
>> *Jan 15 20:19:35.190: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>
>> Sent:
>>
>> SIP/2.0 100 Trying
>>
>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c
>>
>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>
>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>
>> Date: Tue, 15 Jan 2013 20:19:35 GMT
>>
>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>
>> CSeq: 104 INVITE
>>
>> Allow-Events: telephone-event
>>
>> Server: Cisco-SIPGateway/IOS-12.x
>>
>> Content-Length: 0
>>
>> *Jan 15 20:19:35.194: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>
>> Received:
>>
>> SIP/2.0 100 Trying
>>
>> Via: SIP/2.0/UDP 98.192.104.214:5060
>> ;branch=z9hG4bK69232672;received=98.192.104.214
>>
>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>> >;tag=2E6BC0B0-2268
>>
>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>
>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>
>> CSeq: 105 INVITE
>>
>> Server: Asterisk PBX 1.6.2.13
>>
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
>>
>> Supported: replaces, timer
>>
>> Require: timer
>>
>> Session-Expires: 1800;refresher=uas
>>
>> Contact: <sip:17705439047 at 64.154.41.150>
>>
>> Content-Length: 0
>>
>> *Jan 15 20:19:35.194: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>
>> Received:
>>
>> SIP/2.0 200 OK
>>
>> Via: SIP/2.0/UDP 98.192.104.214:5060
>> ;branch=z9hG4bK69232672;received=98.192.104.214
>>
>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>> >;tag=2E6BC0B0-2268
>>
>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>
>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>
>> CSeq: 105 INVITE
>>
>> Server: Asterisk PBX 1.6.2.13
>>
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
>>
>> Supported: replaces, timer
>>
>> Require: timer
>>
>> Session-Expires: 1800;refresher=uas
>>
>> Contact: <sip:17705439047 at 64.154.41.150>
>>
>> Content-Type: application/sdp
>>
>> Content-Length: 333
>>
>> v=0
>>
>> o=root 1685873050 1685873054 IN IP4 64.154.41.150
>>
>> s=Asterisk PBX 1.6.2.13
>>
>> c=IN IP4 64.154.41.150
>>
>> t=0 0
>>
>> m=audio 13014 RTP/AVP 3 8 0 18 101
>>
>> a=rtpmap:3 GSM/8000
>>
>> a=rtpmap:8 PCMA/8000
>>
>> a=rtpmap:0 PCMU/8000
>>
>> a=rtpmap:18 G729/8000
>>
>> a=fmtp:18 annexb=no
>>
>> a=rtpmap:101 telephone-event/8000
>>
>> a=fmtp:101 0-16
>>
>> a=ptime:20
>>
>> a=inactive
>>
>> *Jan 15 20:19:35.198: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>
>> Sent:
>>
>> SIP/2.0 200 OK
>>
>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c
>>
>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>
>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>
>> Date: Tue, 15 Jan 2013 20:19:35 GMT
>>
>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>
>> CSeq: 104 INVITE
>>
>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>>
>> Allow-Events: telephone-event
>>
>> Remote-Party-ID: <sip:17705439047 at 10.1.200.1
>> >;party=called;screen=no;privacy=off
>>
>> Contact: <sip:17705439047 at 10.1.200.1:5060;transport=tcp>
>>
>> Supported: replaces
>>
>> Supported: sdp-anat
>>
>> Server: Cisco-SIPGateway/IOS-12.x
>>
>> Session-Expires: 1800;refresher=uas
>>
>> Require: timer
>>
>> Supported: timer
>>
>> Content-Type: application/sdp
>>
>> Content-Length: 277
>>
>> v=0
>>
>> o=CiscoSystemsSIP-GW-UserAgent 4444 5481 IN IP4 10.1.200.1
>>
>> s=SIP Call
>>
>> c=IN IP4 10.1.200.1
>>
>> t=0 0
>>
>> m=audio 19514 RTP/AVP 0 101 19
>>
>> c=IN IP4 10.1.200.1
>>
>> a=inactive
>>
>> a=rtpmap:0 PCMU/8000
>>
>> a=rtpmap:101 telephone-event/8000
>>
>> a=fmtp:101 0-16
>>
>> a=rtpmap:19 CN/8000
>>
>> a=ptime:20
>>
>> *Jan 15 20:19:35.206: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>
>> Received:
>>
>> ACK sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0
>>
>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28f6dca6616
>>
>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>
>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>
>> Date: Tue, 15 Jan 2013 19:57:42 GMT
>>
>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>
>> Max-Forwards: 70
>>
>> CSeq: 104 ACK
>>
>> Allow-Events: presence, kpml
>>
>> Content-Type: application/sdp
>>
>> Content-Length: 243
>>
>> v=0
>>
>> o=CiscoSystemsCCM-SIP 7322 5 IN IP4 10.1.80.10
>>
>> s=SIP Call
>>
>> c=IN IP4 10.1.10.18
>>
>> b=TIAS:64000
>>
>> b=AS:64
>>
>> t=0 0
>>
>> m=audio 21476 RTP/AVP 0 101
>>
>> a=rtpmap:0 PCMU/8000
>>
>> a=ptime:20
>>
>> a=inactive
>>
>> a=rtpmap:101 telephone-event/8000
>>
>> a=fmtp:101 0-15
>>
>> *Jan 15 20:19:35.210: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>
>> Sent:
>>
>> ACK sip:17705439047 at 64.154.41.150:5060 SIP/2.0
>>
>> Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69246AB
>>
>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>> >;tag=2E6BC0B0-2268
>>
>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>
>> Date: Tue, 15 Jan 2013 20:19:35 GMT
>>
>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>
>> Max-Forwards: 70
>>
>> CSeq: 105 ACK
>>
>> Authorization: Digest username="6784563290",realm="asterisk",uri="
>> sip:17705439047 at 64.154.41.150:5060
>> ",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5
>>
>> Allow-Events: telephone-event
>>
>> Content-Type: application/sdp
>>
>> Content-Length: 265
>>
>> v=0
>>
>> o=CiscoSystemsSIP-GW-UserAgent 3168 2741 IN IP4 98.192.104.214
>>
>> s=SIP Call
>>
>> c=IN IP4 98.192.104.214
>>
>> t=0 0
>>
>> m=audio 19458 RTP/AVP 0 101
>>
>> c=IN IP4 98.192.104.214
>>
>> a=inactive
>>
>> a=rtpmap:0 PCMU/8000
>>
>> a=rtpmap:101 telephone-event/8000
>>
>> a=fmtp:101 0-16
>>
>> a=ptime:20
>>
>> Cisco3825#
>>
>>
>> On Tue, Jan 15, 2013 at 2:28 PM, Ryan Ratliff <rratliff at cisco.com> wrote:
>>
>>> ccsip message is what I'd go with just to see the signaling with no
>>> other stuff. Depending on what that shows and what your gateway is doing
>>> to the signals you may need to expand from there.
>>>
>>> -Ryan
>>>
>>> On Jan 15, 2013, at 2:11 PM, Dane Newman <dane.newman at gmail.com> wrote:
>>>
>>> Ryan
>>>
>>> What is the proper debug to use to caputre the useful information?
>>>
>>> Dane
>>>
>>>
>>>
>>> On Tue, Jan 15, 2013 at 12:42 PM, Ryan Ratliff <rratliff at cisco.com>wrote:
>>>
>>>> Without sip messages I can't get any clues from that.
>>>>
>>>> -Ryan
>>>>
>>>> On Jan 15, 2013, at 12:35 PM, Dane Newman <dane.newman at gmail.com>
>>>> wrote:
>>>>
>>>> Thanks Ryan for the input
>>>>
>>>>
>>>> *On the call when I hold the call the following debug pops out....*
>>>>
>>>>
>>>> *Jan 15 17:56:05.246:
>>>> //13939/922252E78D73/SIP/Error/ccsip_api_request_offer: Unable to add
>>>> passthru hdrs to
>>>> container
>>>> SIP: Attribute mid, level 1 instance 1 not found.
>>>> SIP: (13938) Group (a= group line) attribute, level 65535 instance 1
>>>> not found.
>>>> *Jan 15 17:56:05.274:
>>>> //13938/922252E78D73/SIP/Error/ccsip_api_response_answer: Unable to add
>>>> passthru headers to container
>>>> SIP: Attribute mid, level 1 instance 1 not found.
>>>> SIP: (13939) Group (a= group line) attribute, level 65535 instance 1
>>>> not found.
>>>> *Jan 15 17:56:05.286:
>>>> //13939/922252E78D73/SIP/Error/ccsip_api_request_offer: Unable to add
>>>> passthru hdrs to
>>>> container
>>>> *Jan 15 17:56:05.302:
>>>> //13938/922252E78D73/SIP/Error/ccsip_api_response_answer: Unable to add
>>>> passthru headers to container
>>>> SIP: Attribute mid, level 1 instance 1 not found.
>>>> SIP: (13939) Group (a= group line) attribute, level 65535 instance 1
>>>> not found.
>>>> SIP: Attribute mid, level 1 instance 1 not found.
>>>> *Jan 15 17:56:05.322:
>>>> //13939/922252E78D73/SIP/Error/sipSPIProcessAckMedia: Could not modify QoS
>>>> params for midcall INVITE
>>>>
>>>> *After I try to unhold the call the following debug comes out....*
>>>> **
>>>>
>>>> *Jan 15 17:56:18.874:
>>>> //13939/922252E78D73/SIP/Error/ccsip_api_request_offer: Unable to add
>>>> passthru hdrs to
>>>> container
>>>> *Jan 15 17:56:18.894:
>>>> //13938/922252E78D73/SIP/Error/ccsip_api_response_answer: Unable to add
>>>> passthru headers to container
>>>> SIP: Attribute mid, level 1 instance 1 not found.
>>>> SIP: (13939) Group (a= group line) attribute, level 65535 instance 1
>>>> not found.
>>>> SIP: Attribute mid, level 1 instance 1 not found.
>>>> *Jan 15 17:56:18.906:
>>>> //13939/922252E78D73/SIP/Error/sipSPIProcessAckMedia: Could not modify QoS
>>>> params for midcall INVITE
>>>> Cisco3825#
>>>> Cisco3825#
>>>> Cisco3825#
>>>>
>>>> On Tue, Jan 15, 2013 at 9:42 AM, Ryan Ratliff <rratliff at cisco.com>wrote:
>>>>
>>>>> Given you have an ITSP it's most likely the initial hold that's
>>>>> failing, which is only manifesting when you try to resume it. If you
>>>>> haven't noticed already this is also very likely causing transfers to fail.
>>>>>
>>>>> Take a look at the SIP signaling for a call. I believe the most
>>>>> common cause to this is the ITSP not handling our transition from
>>>>> active->inactive->sendonly->active from hold to MOH to resume. The
>>>>> "Duplex Streaming Enabled" parameter is there just for this type of problem.
>>>>>
>>>>> -Ryan
>>>>>
>>>>> On Jan 14, 2013, at 6:40 PM, Dane Newman <dane.newman at gmail.com>
>>>>> wrote:
>>>>>
>>>>> *Hello Kenneth*
>>>>> **
>>>>> *I have restarted both CUCM servers so this should have restarted the
>>>>> services when the utils system restart happened*
>>>>> **
>>>>>
>>>>> *on my router I see I am using g711 from the debug *
>>>>> **
>>>>> *I ran a debug voip ccapi inout *
>>>>> **
>>>>> *I connected a call calling from an external number to a DiD inside
>>>>> of my system. Once the call was connected I put the call on hold and the
>>>>> following debug came out..the music on hold played for the external caller
>>>>> *
>>>>>
>>>>> *Jan 14 23:47:40.779: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:
>>>>> Stop Tone On Digit=FALSE, Tone=Null,
>>>>> Tone Direction=Sum Network, Params=0x0, Call Id=12741
>>>>> *Jan 14 23:47:40.783:
>>>>> //12741/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
>>>>> Call Id=12742, Xmit Function=0x64204BAC
>>>>> *Jan 14 23:47:40.783:
>>>>> //12741/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
>>>>> *Jan 14 23:47:40.783: cc_api_get_xcode_stream : 4702
>>>>> *Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
>>>>> Call Id=12742,
>>>>> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>>>>> Modem=0x0, Codec Bytes=20, Signal Type=2)
>>>>> *Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
>>>>> Playout Max=1000(ms), Fax Nom=300(ms))
>>>>> *Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
>>>>> Call Id=12741,
>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)
>>>>> *Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
>>>>> Call Id=12741,
>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)
>>>>> *Jan 14 23:47:40.783:
>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>> Event=170, Call Id=12742
>>>>> *Jan 14 23:47:40.783:
>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>> Event Is Sent To Conferenced SPI(s) Directly
>>>>> *Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_call_feature:
>>>>> Feature Type=50, Interface=0xC05A65AC, Call Id=12742
>>>>> *Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
>>>>> Call Id=12741,
>>>>> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>>>>> Modem=0x0, Codec Bytes=20, Signal Type=2)
>>>>> *Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
>>>>> Playout Max=1000(ms), Fax Nom=300(ms))
>>>>> *Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
>>>>> Call Id=12742,
>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
>>>>> *Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
>>>>> Call Id=12742,
>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
>>>>> *Jan 14 23:47:40.811:
>>>>> //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>> Event=171, Call Id=12741
>>>>> *Jan 14 23:47:40.811:
>>>>> //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>> Event Is Sent To Conferenced SPI(s) Directly
>>>>> *Jan 14 23:47:40.815: //12742/9906C1828C05/CCAPI/cc_api_call_facility:
>>>>> Interface=0xC05A65AC, Call Id=12742
>>>>> *Jan 14 23:47:40.819: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:
>>>>> Stop Tone On Digit=FALSE, Tone=Null,
>>>>> Tone Direction=Sum Network, Params=0x0, Call Id=12741
>>>>> *Jan 14 23:47:40.819:
>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>> Event=96, Call Id=12742
>>>>> *Jan 14 23:47:40.819:
>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>> Event Is Sent To Conferenced SPI(s) Directly
>>>>> *Jan 14 23:47:40.839:
>>>>> //12742/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
>>>>> Call Id=12741, Xmit Function=0x64204BAC
>>>>> *Jan 14 23:47:40.839:
>>>>> //12742/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
>>>>> *Jan 14 23:47:40.839: cc_api_get_xcode_stream : 4702
>>>>> *Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
>>>>> Call Id=12741,
>>>>> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>>>>> Modem=0x0, Codec Bytes=20, Signal Type=2)
>>>>> *Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
>>>>> Playout Max=1000(ms), Fax Nom=300(ms))
>>>>> *Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
>>>>> Call Id=12742,
>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
>>>>> *Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
>>>>> Call Id=12742,
>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
>>>>> *Jan 14 23:47:40.843:
>>>>> //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>> Event=170, Call Id=12741
>>>>> *Jan 14 23:47:40.843:
>>>>> //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>> Event Is Sent To Conferenced SPI(s) Directly
>>>>> *Jan 14 23:47:40.859: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
>>>>> Call Id=12742,
>>>>> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>>>>> Modem=0x0, Codec Bytes=20, Signal Type=2)
>>>>> *Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
>>>>> Playout Max=1000(ms), Fax Nom=300(ms))
>>>>> *Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
>>>>> Call Id=12741,
>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)
>>>>> *Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
>>>>> Call Id=12741,
>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)
>>>>> *Jan 14 23:47:40.863:
>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>> Event=171, Call Id=12742
>>>>> *Jan 14 23:47:40.863:
>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>> Event Is Sent To Conferenced SPI(s) Directly
>>>>> *Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_call_facility:
>>>>> Interface=0xC05A65AC, Call Id=12742
>>>>> Cisco3825#
>>>>> Cisco3825#
>>>>> Cisco3825#
>>>>>
>>>>>
>>>>> *I then after that took off the hold and the following debug came
>>>>> out. The call on the PSDN side still played the hold music while there was
>>>>> no voice on the deskphone side.*
>>>>>
>>>>> *Jan 14 23:47:40.779: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:
>>>>> Stop Tone On Digit=FALSE, Tone=Null,
>>>>> Tone Direction=Sum Network, Params=0x0, Call Id=12741
>>>>> *Jan 14 23:47:40.783:
>>>>> //12741/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
>>>>> Call Id=12742, Xmit Function=0x64204BAC
>>>>> *Jan 14 23:47:40.783:
>>>>> //12741/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
>>>>> *Jan 14 23:47:40.783: cc_api_get_xcode_stream : 4702
>>>>> *Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
>>>>> Call Id=12742,
>>>>> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>>>>> Modem=0x0, Codec Bytes=20, Signal Type=2)
>>>>> *Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
>>>>> Playout Max=1000(ms), Fax Nom=300(ms))
>>>>> *Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
>>>>> Call Id=12741,
>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)
>>>>> *Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
>>>>> Call Id=12741,
>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)
>>>>> *Jan 14 23:47:40.783:
>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>> Event=170, Call Id=12742
>>>>> *Jan 14 23:47:40.783:
>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>> Event Is Sent To Conferenced SPI(s) Directly
>>>>> *Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_call_feature:
>>>>> Feature Type=50, Interface=0xC05A65AC, Call Id=12742
>>>>> *Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
>>>>> Call Id=12741,
>>>>> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>>>>> Modem=0x0, Codec Bytes=20, Signal Type=2)
>>>>> *Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
>>>>> Playout Max=1000(ms), Fax Nom=300(ms))
>>>>> *Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
>>>>> Call Id=12742,
>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
>>>>> *Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
>>>>> Call Id=12742,
>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
>>>>> *Jan 14 23:47:40.811:
>>>>> //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>> Event=171, Call Id=12741
>>>>> *Jan 14 23:47:40.811:
>>>>> //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>> Event Is Sent To Conferenced SPI(s) Directly
>>>>> *Jan 14 23:47:40.815: //12742/9906C1828C05/CCAPI/cc_api_call_facility:
>>>>> Interface=0xC05A65AC, Call Id=12742
>>>>> *Jan 14 23:47:40.819: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:
>>>>> Stop Tone On Digit=FALSE, Tone=Null,
>>>>> Tone Direction=Sum Network, Params=0x0, Call Id=12741
>>>>> *Jan 14 23:47:40.819:
>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>> Event=96, Call Id=12742
>>>>> *Jan 14 23:47:40.819:
>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>> Event Is Sent To Conferenced SPI(s) Directly
>>>>> *Jan 14 23:47:40.839:
>>>>> //12742/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
>>>>> Call Id=12741, Xmit Function=0x64204BAC
>>>>> *Jan 14 23:47:40.839:
>>>>> //12742/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
>>>>> *Jan 14 23:47:40.839: cc_api_get_xcode_stream : 4702
>>>>> *Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
>>>>> Call Id=12741,
>>>>> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>>>>> Modem=0x0, Codec Bytes=20, Signal Type=2)
>>>>> *Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
>>>>> Playout Max=1000(ms), Fax Nom=300(ms))
>>>>> *Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
>>>>> Call Id=12742,
>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
>>>>> *Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
>>>>> Call Id=12742,
>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)
>>>>> *Jan 14 23:47:40.843:
>>>>> //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>> Event=170, Call Id=12741
>>>>> *Jan 14 23:47:40.843:
>>>>> //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>> Event Is Sent To Conferenced SPI(s) Directly
>>>>> *Jan 14 23:47:40.859: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source
>>>>> Call Id=12742,
>>>>> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>>>>> Modem=0x0, Codec Bytes=20, Signal Type=2)
>>>>> *Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
>>>>> Playout Max=1000(ms), Fax Nom=300(ms))
>>>>> *Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
>>>>> Call Id=12741,
>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)
>>>>> *Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source
>>>>> Call Id=12741,
>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),
>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)
>>>>> *Jan 14 23:47:40.863:
>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>> Event=171, Call Id=12742
>>>>> *Jan 14 23:47:40.863:
>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>> Event Is Sent To Conferenced SPI(s) Directly
>>>>> *Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_call_facility:
>>>>> Interface=0xC05A65AC, Call Id=12742
>>>>> Cisco3825#
>>>>> Cisco3825#
>>>>> Cisco3825#
>>>>>
>>>>> On Mon, Jan 14, 2013 at 6:20 PM, Kenneth Hayes <
>>>>> kennethwhayes at gmail.com> wrote:
>>>>>
>>>>>> Have you also restarted the Cisco IP Media Services?
>>>>>>
>>>>>> Sent from my iPhone
>>>>>>
>>>>>> On Jan 14, 2013, at 6:12 PM, Dane Newman <dane.newman at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>> My ITSP will only support 711ulaw for me currently I believe. They
>>>>>> hard coded it with me when I was initially setting it up.
>>>>>>
>>>>>> Do you think this could be a codec issue? How would I go about
>>>>>> identifying if it is?
>>>>>>
>>>>>> Dane
>>>>>>
>>>>>> On Mon, Jan 14, 2013 at 6:09 PM, Kenneth Hayes <
>>>>>> kennethwhayes at gmail.com> wrote:
>>>>>>
>>>>>>> Have you tried different audio codecs?
>>>>>>>
>>>>>>> Sent from my iPhone
>>>>>>>
>>>>>>> On Jan 14, 2013, at 6:06 PM, Dane Newman <dane.newman at gmail.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>> Ryan (sorry I forgot to reply to all)
>>>>>>>
>>>>>>> Thanks for the Reply
>>>>>>> Oddly enough we are.
>>>>>>> This probably has something to do with MOH in general?
>>>>>>>
>>>>>>> Internally when I user puts another user on hold everything works.
>>>>>>> No MOH plays and they can hold and unhold the call just fine.
>>>>>>> I tested calling from an external number. Once I put the external
>>>>>>> caller on hold the MOH played but I was unable to resume the call. When I
>>>>>>> hit resume on the deskphone the MOH still played to the external caller and
>>>>>>> there was no sound on the deskphone.
>>>>>>>
>>>>>>> On Mon, Jan 14, 2013 at 5:25 PM, Ryan Ratliff <rratliff at cisco.com>wrote:
>>>>>>>
>>>>>>>> Do you get similar behavior if you just hold and resume the call
>>>>>>>> outside SNR features?
>>>>>>>>
>>>>>>>> -Ryan
>>>>>>>>
>>>>>>>> On Jan 14, 2013, at 4:18 PM, Dane Newman <dane.newman at gmail.com>
>>>>>>>> wrote:
>>>>>>>>
>>>>>>>> Using keyboard-interactive authentication.
>>>>>>>>
>>>>>>>> Password:
>>>>>>>>
>>>>>>>>
>>>>>>>> Cisco3825#
>>>>>>>>
>>>>>>>> Cisco3825#sh ver
>>>>>>>>
>>>>>>>> Cisco IOS Software, 3800 Software (C3825-ADVENTERPRISEK9_IVS_LI-M),
>>>>>>>> Version 15.1
>>>>>>>> (4)M5, RELEASE SOFTWARE (fc1)
>>>>>>>>
>>>>>>>> Technical Support: http://www.cisco.com/techsupport
>>>>>>>> Copyright (c) 1986-2012 by Cisco Systems, Inc.
>>>>>>>>
>>>>>>>> Compiled Tue 04-Sep-12 17:25 by prod_rel_team
>>>>>>>>
>>>>>>>>
>>>>>>>> ROM: System Bootstrap, Version 12.4(13r)T10, RELEASE SOFTWARE (fc1)
>>>>>>>>
>>>>>>>>
>>>>>>>> Cisco3825 uptime is 1 week, 1 day, 1 hour, 38 minutes
>>>>>>>>
>>>>>>>> System returned to ROM by power-on
>>>>>>>>
>>>>>>>> System image file is
>>>>>>>> "flash:c3825-adventerprisek9_ivs_li-mz.151-4.M5.bin"
>>>>>>>> Last reload type: Normal Reload
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> This product contains cryptographic features and is subject to
>>>>>>>> United
>>>>>>>> States and local country laws governing import, export, transfer
>>>>>>>> and
>>>>>>>> use. Delivery of Cisco cryptographic products does not imply
>>>>>>>>
>>>>>>>> third-party authority to import, export, distribute or use
>>>>>>>> encryption.
>>>>>>>> Importers, exporters, distributors and users are responsible for
>>>>>>>>
>>>>>>>> compliance with U.S. and local country laws. By using this product
>>>>>>>> you
>>>>>>>> agree to comply with applicable laws and regulations. If you are
>>>>>>>> unable
>>>>>>>> to comply with U.S. and local laws, return this product
>>>>>>>> immediately.
>>>>>>>>
>>>>>>>> A summary of U.S. laws governing Cisco cryptographic products may
>>>>>>>> be found at:
>>>>>>>> http://www.cisco.com/wwl/export/crypto/tool/stqrg.html
>>>>>>>>
>>>>>>>> If you require further assistance please contact us by sending
>>>>>>>> email to
>>>>>>>> export at cisco.com.
>>>>>>>>
>>>>>>>>
>>>>>>>> Cisco 3825 (revision 1.2) with 1011712K/36864K bytes of memory.
>>>>>>>>
>>>>>>>> Processor board ID FTX1237A1T0
>>>>>>>>
>>>>>>>> 2 Gigabit Ethernet interfaces
>>>>>>>>
>>>>>>>> 2 Channelized T1/PRI ports
>>>>>>>>
>>>>>>>> 1 Virtual Private Network (VPN) Module
>>>>>>>>
>>>>>>>> DRAM configuration is 64 bits wide with parity enabled.
>>>>>>>>
>>>>>>>> 479K bytes of NVRAM.
>>>>>>>>
>>>>>>>> 500472K bytes of ATA System CompactFlash (Read/Write)
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> License Info:
>>>>>>>>
>>>>>>>>
>>>>>>>> License UDI:
>>>>>>>>
>>>>>>>>
>>>>>>>> -------------------------------------------------
>>>>>>>>
>>>>>>>> Device# PID SN
>>>>>>>>
>>>>>>>> Sent from my mobile device
>>>>>>>>
>>>>>>>> On Jan 14, 2013, at 4:11 PM, Kenneth Hayes <kennethwhayes at gmail.com>
>>>>>>>> wrote:
>>>>>>>>
>>>>>>>> What version of code are you running on the CUBE?
>>>>>>>>
>>>>>>>> Sent from my iPhone
>>>>>>>>
>>>>>>>> On Jan 14, 2013, at 3:43 PM, Dane Newman <dane.newman at gmail.com>
>>>>>>>> wrote:
>>>>>>>>
>>>>>>>> Hello
>>>>>>>>
>>>>>>>> I have an issue when users are connected to a call and hit the
>>>>>>>> mobility soft key button on 9971 phones when a call is active, the phone
>>>>>>>> system rings on the mobile number configured in the system. When they pick
>>>>>>>> up the the mobile number it just plays what sounds like hold music on both
>>>>>>>> ends of the call (I believe this music is coming from cucm but I haven't
>>>>>>>> heard it before) instead of providing 2 way voice.
>>>>>>>>
>>>>>>>> In another senario with what I believe is the same issue. If a user
>>>>>>>> picks up on there cell phone first (using single number reach) opposed to
>>>>>>>> the deskphone the call is connected with 2 way voice and no issues exist.
>>>>>>>> If the user then hangs up his cell phone with the intent to take the call
>>>>>>>> on his deskphone the calling party starts hearing the hold music. Once the
>>>>>>>> user picks up the call on his deskphone he hears nothing but the calling
>>>>>>>> party is still hearing the hold music. It is not working as intended where
>>>>>>>> 2 way voice happens once the user hangs up his mobile phone and picks up on
>>>>>>>> his deskphone 2 way voice should happen.
>>>>>>>>
>>>>>>>> My topology is as follows..
>>>>>>>>
>>>>>>>>
>>>>>>>> PSDN --> SIP TRUNK FROM ITSP --> 3825 CUBE --->CUCM -->DESKPHOHE
>>>>>>>>
>>>>>>>> Calls are sent back out the SIP trunk to the ITSP when using mobile
>>>>>>>> connect/snr.
>>>>>>>>
>>>>>>>> Does anyone have any ideas how I can make 2 way voice happen
>>>>>>>> instead of the hold music when the calls are picked up?
>>>>>>>> _______________________________________________
>>>>>>>> cisco-voip mailing list
>>>>>>>> cisco-voip at puck.nether.net
>>>>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> cisco-voip mailing list
>>>>>>>> cisco-voip at puck.nether.net
>>>>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>
>>>
>>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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