[cisco-voip] Mobility Issue
Kenneth Hayes
kennethwhayes at gmail.com
Tue Jan 15 15:42:17 EST 2013
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
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>
>>>>
>>>
>>>
>>
>>
>
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