[cisco-voip] Mobility Issue
Kenneth Hayes
kennethwhayes at gmail.com
Tue Jan 15 15:31:45 EST 2013
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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