Pick Up Call on Hold with Pickup Softkey

On CME if you put a call on hold on your extension, you can use the call pick and dial the extension number and it will pull the call to another phone.
Does CUCM have that featue with out parking the call?
I want to be able to put a call on hold...walk up to a different phone and pull that call to that phone
Everthing I read about Call PickUp, Directed Call PickUp, Group Pick up is for ringing calls not for calls on hold.

Have you looked at Directed Call Pickup feature, I am unsure if it can pickup a held call or just ringing call though. Might be something you would want to test.
HTH,
Chris

Similar Messages

Is there a way to put an incoming 2nd call on hold without answering it while I finish my first call

Years ago I had a feature on a phone that when I was on one call and another call beeped in, I could just hit a button and it would put the second call on hold with a message that I was just finishing another call and to please hold. Then, when I finished with the first call, I just switch or hang up.
I am constantly losing calls because I can't get off the first one quick enough to answer the second call.
Maybe there is an app for that?

Guys, guys... (sigh)
Now, let's be constructive, shall we...?
Malcolm, I see you live in a country even colder than mine, so maybe you're speaking from experience!
No, seriously, wjosten put it plainly: save from jailbreaking my phone, there doesn't seem to be a solution to my "problem". I'm not quite ready to jailbreak my phone yet, but I'm getting closer every day. I love that phone, but there are so many built-in limitations that I'm on the brink of doing it.
Now, this said, it would make for a nice improvement in future versions of the iPhone: let the user set one of the physical controls to take calls.
(As for buying those gloves you guys mention, I like my wool gloves just fine. Plus they're a gift from a loved one)

Cannot pick up calls on wait, Cannot make a second call holding the first call

I have a really weird problem with my iPhone 5S. (iOS 7.1)
1. When I am on a call, I am not able to pick up a second call that is on wait. I can see the second call, I can hear the beep sounds but as soon as I pick up, the call freezes in 00.00 time, and My first caller is still active, not on hold.
2. When I am on a call, I am not able to make a second call placing the first call on hold. The second call disconnects immediately with 3 beeps.
3. I am able to see the incoming calls on wait, when I am already on a call. I am able to hear the beeps coming in. SO it means the call waiting is on (Tried turning it on/off/on too.) But when I pick up the call, the above problem comes in.
What have I tried to solve this: (None of them helped)
1. I tried a hard reset of my iPhone (Holding Power and Home button)
2. I tried toggling the Call waiting button
3. I tried resetting network settings.
4. I called my Operator for assistance. I am using Airtel, India. They told me to dial this code - *43# . I tried it and it gave me this screen. But the problem is not solved yet.
Is there anybody here who is facing the same issue with your iPhone? I never had this problem with my iPhone 4.

Make sure you do not have "Do Not Disturb" turned on. This feature will block the first call, but is someone calls right back it will be put through.
On the iPhone, go to Settings/Do Not Disturb
Make sure you do not have Do Not Disturb turned on manual or scheduled.

Pick and Pack / Goods Issue with Multi Batch Pickup

Dear Friends,
Can anyone explain me the process of Pick and Pack / Goods Issue with Multi Batch Pickup.
Harsh

Hi Mathur,
Please refer these links on working with batch splits in deliveries :
<a href="http://help.sap.com/saphelp_47x200/helpdata/en/38/c1853488601e33e10000009b38f83b/frameset.htm">Batch split</a>
<a href="http://help.sap.com/saphelp_47x200/helpdata/en/38/c1853488601e33e10000009b38f83b/frameset.htm">Executing Multi batch</a>
<a href="http://help.sap.com/saphelp_47x200/helpdata/en/38/c1853488601e33e10000009b38f83b/frameset.htm">Transferring Batch Data from Deliveries into Warehouse Management</a>
Hope it helps. Please reward if useful.
Thanks & Regards
Sadhu Kishore

Not able to pick up calls without my headphones with iphone 4....what to do?

I have an iphone 4. I listened to my music with the ipod app. Now I am unable to pick up calls without using the headphones. The sound seems to be locked somehow. How do I fix this?

Try plugging and unplugging your earphone a couple of times and see it that fixes your issue. Sometimes the headphone switch does not switch the audio back to the phone.

Avaya calls to CUCM 8.6: Call Park/Hold Fail

This may be similar to the issue I posted here: https://supportforums.cisco.com/message/3496110#3496110
4 digit dialing from an Avaya system to a CUCM 8.6.2 server works fine, the Avaya is set as an h.323 gateway in CUCM.
Once the call is active if the Cisco phone places the Avaya caller on hold they hear no hold music and the call stays active until the Cisco phone tries to retrieve it. Then it's a fast busy on both end.
The same thing happens with calls placed on park.
Any suggestions?

Hi
Are you using G711 or G729 for these calls? If using G729 the CUCM software MTP cannot be used as it only supports g711, you would have to use a gateway-based MTP.
Try setting the calls to G711 to see if it works as a test.
Aaron

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

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

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

Music When Putting Call In Hold

I want to ask how many people would like the opportunity to replace that annoying busy signal that is generated when a call is put on hold with music more enjoyable ..?
Not be a revolutionary idea for Apple iPhone and phones? Since there is a command of "call waiting" because it can not replace an ugly busy signal repeater with music?
:-D might be the idea of a new and unique rinvoluzionario updated version of the software for the iPhone.

It'd be impossible for Apple to implement it on the phone, as the tones are a network (carrier) service/feature and not generated on the phone at all.
All the ringing, busy, and on-hold sounds that a caller may hear, come from the network, not the recipients phone.
And, as mentioned, some networks already have services where you can change those tones (for everyone calling or even different sounds for different callers).

Calls being rejected with CTI failure / timeout

I have UCCX 8.5.1.11002-22 deployed with UCM 8.5.1.13900-5.
This is a new installation.
The site is experiencing a high number of Rejected calls. I getting a mixture of two errors:
ABLISHED]->HELD; nested exception is: com.cisco.jtapi.PlatformExceptionImpl: Cti request timed out
and
Reject: Trigger Failed
When this starts almost 50% of calls hit this error, and is only recoverable by restarting the UCCX Engine. I have synch'd the JTAPI versions and no changes have been made. I have also checked the checked CTI Ports and CTI Route points and all are registered and configured.
Has anyone else seen a similar issue with UCCX 8.5?
Regards,
Alex

On scrolling through the logs I see a high number of the CTI timeouts when attempting to take the call off hold (while in the queue)
I noticed that the MoH Media Server had Multi-casting enabled. I have disabled. Does anyone know if this could potentially be related as I believe multi-casting is not supported with UCCX.
Thanks,
Alex

Plz tell me BDC CALL TRANSACTION steps with simple example

hi,
plz tell me the steps
BDC CALL TRANSACTION steps with simple example

Hi,
BATCH DATA COMMUNICATION
About Data Transfer In R/3 System
When a company decides to implement the SAP R/3 to manage business-critical data, it usually does not start from a no-data situation. Normally, a SAP R/3 project comes into replace or complement existing application.
In the process of replacing current applications and transferring application data, two situations might occur:
? The first is when application data to be replaced is transferred at once, and only once.
? The second situation is to transfer data periodically from external systems to SAP and vice versa.
? There is a period of time when information has to be transferred from existing application, to SAP R/3, and often this process will be repetitive.
The SAP system offers two primary methods for transferring data into SAP systems. From non-SAP systems or legacy system. These two methods are collectively called ?batch input? or ?batch data communication?.
1. SESSION METHOD
2. CALL TRANSACTION
3. DIRECT INPUT
Advantages offered by BATCH INPUT method:
1. Can process large data volumes in batch.
2. Can be planned and submitted in the background.
3. No manual interaction is required when data is transferred.
4. Data integrity is maintained as whatever data is transferred to the table is through transaction. Hence batch input data is submitted to all the checks and validations.
To implement one of the supported data transfers, you must often write the program that exports the data from your non-SAP system. This program, known as a ?data transfer? program must map the data from the external system into the data structure required by the SAP batch input program.
The batch input program must build all of the input to execute the SAP transaction.
Two main steps are required:
? To build an internal table containing every screen and every field to be filled in during the execution of an SAP transaction.
? To pass the table to SAP for processing.
Prerequisite for Data Transfer Program
Writing a Data Transfer Program involves following prerequisites:
Analyzing data from local file
Analyzing transaction
Analyzing transaction involves following steps:
? The transaction code, if you do not already know it.
? Which fields require input i.e., mandatory.
? Which fields can you allow to default to standard values.
? The names, types, and lengths of the fields that are used by a transaction.
? Screen number and Name of module pool program behind a particular transaction.
To analyze a transaction::
? Start the transaction by menu or by entering the transaction code in the command box.
(You can determine the transaction name by choosing System ? Status.)
? Step through the transaction, entering the data will be required for processing your batch input data.
? On each screen, note the program name and screen (dynpro) number.
(dynpro = dyn + pro. Dyn = screen, pro = number)
? Display these by choosing System ? Status. The relevant fields are Program (dynpro) and Dynpro number. If pop-up windows occur during execution, you can get the program name and screen number by pressing F1 on any field or button on the screen.
The technical info pop-up shows not only the field information but also the program and screen.
? For each field, check box, and radio button on each screen, press F1 (help) and then choose Technical Info.
Note the following information:
- The field name for batch input, which you?ll find in its own box.
- The length and data type of the field. You can display this information by double clicking on the Data Element field.
? Find out the identification code for each function (button or menu) that you must execute to process the batch-input data (or to go to new screen).
Place the cursor on the button or menu entry while holding down the left mouse button. Then press F1.
In the pop-up window that follows, choose Technical info and note the code that is shown in the Function field.
You can also run any function that is assigned to a function key by way of the function key number. To display the list of available function keys, click on the right mouse button. Note the key number that is assigned to the functions you want to run.
Once you have program name, screen number, field name (screen field name), you can start writing.
DATA TRANSFER program.
Declaring internal table
First Integral Table similar to structure like local file.
Declaring internal table like BDCDATA
The data from internal table is not transferred directly to database table, it has to go through transaction. You need to pass data to particular screen and to particular screen-field. Data is passed to transaction in particular format, hence there is a need for batch input structure.
The batch input structure stores the data that is to be entered into SAP system and the actions that are necessary to process the data. The batch input structure is used by all of the batch input methods. You can use the same structure for all types of batch input, regardless of whether you are creating a session in the batch input queue or using CALL TRANSACTION.
This structure is BDCDATA, which can contain the batch input data for only a single run of a transaction. The typical processing loop in a program is as follows:
? Create a BDCDATA structure
? Write the structure out to a session or process it with CALL TRANSACTION USING; and then
? Create a BDCDATA structure for the next transaction that is to be processed.
Within a BDCDATA structure, organize the data of screens in a transaction. Each screen that is processed in the course of a transaction must be identified with a BDCDATA record. This record uses the Program, Dynpro, and Dynbegin fields of the structure.
The screen identifier record is followed by a separate BDCDATA record for each value, to be entered into a field. These records use the FNAM and FVAL fields of the BDCDATA structure. Values to be entered in a field can be any of the following:
? Data that is entered into screen fields.
? Function codes that are entered into the command field. Such function codes execute functions in a transaction, such as Save or Enter.
The BDCDATA structure contains the following fields:
? PROGRAM: Name of module pool program associated with the screen. Set this field only for the first record for the screen.
? DYNPRO: Screen Number. Set this field only in the first record for the screen.
? DYNBEGIN: Indicates the first record for the screen. Set this field to X, only for the first record for the screen. (Reset to ? ? (blank) for all other records.)
? FNAM: Field Name. The FNAM field is not case-sensitive.
? FVAL: Value for the field named in FNAM. The FVAL field is case-sensitive. Values assigned to this field are always padded on the right, if they are less than 132 characters. Values must be in character format.
Transferring data from local file to internal table
Data is uploaded to internal table by UPLOAD of WS_UPLOAD function.
Population of BDCDATA
For each record of internal table, you need to populate Internal table, which is similar to BDCDATA structure.
All these five initial steps are necessary for any type of BDC interface.
DATA TRANSFER program can call SESSION METHOD or CALL TRANSACTION. The initial steps for both the methods are same.
First step for both the methods is to upload the data to internal table. From Internal Table, the data is transferred to database table by two ways i.e., Session method and Call transaction.
SESSION METHOD
About Session method
In this method you transfer data from internal table to database table through sessions.
In this method, an ABAP/4 program reads the external data that is to be entered in the SAP System and stores the data in session. A session stores the actions that are required to enter your data using normal SAP transaction i.e., Data is transferred to session which in turn transfers data to database table.
Session is intermediate step between internal table and database table. Data along with its action is stored in session i.e., data for screen fields, to which screen it is passed, the program name behind it, and how the next screen is processed.
When the program has finished generating the session, you can run the session to execute the SAP transactions in it. You can either explicitly start and monitor a session or have the session run in the background processing system.
Unless session is processed, the data is not transferred to database table.
BDC_OPEN_GROUP
You create the session through program by BDC_OPEN_GROUP function.
Parameters to this function are:
? User Name: User name
? Group: Name of the session
? Lock Date: The date on which you want to process the session.
? Keep: This parameter is passed as ?X? when you want to retain session after
processing it or ? ? to delete it after processing.
BDC_INSERT
This function creates the session & data is transferred to Session.
Parameters to this function are:
? Tcode: Transaction Name
? Dynprotab: BDC Data
BDC_CLOSE_GROUP
This function closes the BDC Group. No Parameters.
Some additional information for session processing
When the session is generated using the KEEP option within the BDC_OPEN_GROUP, the system always keeps the sessions in the queue, whether it has been processed successfully or not.
However, if the session is processed, you have to delete it manually. When session processing is completed successfully while KEEP option was not set, it will be removed automatically from the session queue. Log is not removed for that session.
If the batch-input session is terminated with errors, then it appears in the list of INCORRECT session and it can be processed again. To correct incorrect session, you can analyze the session. The Analysis function allows to determine which screen and value has produced the error. If you find small errors in data, you can correct them interactively, otherwise you need to modify batch input program, which has generated the session or many times even the data file.
CALL TRANSACTION
About CALL TRANSACTION
A technique similar to SESSION method, while batch input is a two-step procedure, Call Transaction does both steps online, one after the other. In this method, you call a transaction from your program by
Call transaction <tcode> using <BDCTAB>
Mode <A/N/E>
Update <S/A>
Messages into <MSGTAB>.
Parameter ? 1 is transaction code.
Parameter ? 2 is name of BDCTAB table.
Parameter ? 3 here you are specifying mode in which you execute transaction
A is all screen mode. All the screen of transaction are displayed.
N is no screen mode. No screen is displayed when you execute the transaction.
E is error screen. Only those screens are displayed wherein you have error record.
Parameter ? 4 here you are specifying update type by which database table is updated.
S is for Synchronous update in which if you change data of one table then all the related Tables gets updated. And sy-subrc is returned i.e., sy-subrc is returned for once and all.
A is for Asynchronous update. When you change data of one table, the sy-subrc is returned. And then updating of other affected tables takes place. So if system fails to update other tables, still sy-subrc returned is 0 (i.e., when first table gets updated).
Parameter ? 5 when you update database table, operation is either successful or unsuccessful or operation is successful with some warning. These messages are stored in internal table, which you specify along with MESSAGE statement. This internal table should be declared like BDCMSGCOLL, a structure available in ABAP/4. It contains the following fields:
1. Tcode: Transaction code
2. Dyname: Batch point module name
3. Dynumb: Batch input Dyn number
4. Msgtyp: Batch input message type (A/E/W/I/S)
5. Msgspra: Batch input Lang, id of message
6. Msgid: Message id
7. MsgvN: Message variables (N = 1 - 4)
For each entry, which is updated in database, table message is available in BDCMSGCOLL. As BDCMSGCOLL is structure, you need to declare a internal table which can contain multiple records (unlike structure).
Steps for CALL TRANSACTION method
1. Internal table for the data (structure similar to your local file)
2. BDCTAB like BDCDATA
3. UPLOAD or WS_UPLOAD function to upload the data from local file to itab. (Considering file is local file)
4. Loop at itab.
Populate BDCTAB table.
Call transaction <tcode> using <BDCTAB>
Mode <A/N/E>
Update <S/A>.
Refresh BDCTAB.
Endloop.
(To populate BDCTAB, You need to transfer each and every field)
The major differences between Session method and Call transaction are as follows:
SESSION METHOD CALL TRANSACTION
1. Data is not updated in database table unless Session is processed. Immediate updation in database table.
2. No sy-subrc is returned. Sy-subrc is returned.
3. Error log is created for error records. Errors need to be handled explicitly
4. Updation in database table is always synchronous Updation in database table can be synchronous Or Asynchronous.
Error Handling in CALL TRANSACTION
When Session Method updates the records in database table, error records are stored in the log file. In Call transaction there is no such log file available and error record is lost unless handled. Usually you need to give report of all the error records i.e., records which are not inserted or updated in the database table. This can be done by the following method:
Steps for the error handling in CALL TRANSACTION
1. Internal table for the data (structure similar to your local file)
2. BDCTAB like BDCDATA
3. Internal table BDCMSG like BDCMSGCOLL
4. Internal table similar to Ist internal table
(Third and fourth steps are for error handling)
5. UPLOAD or WS_UPLOAD function to upload the data from the local file to itab. (Considering file is local file)
6. Loop at itab.
Populate BDCTAB table.
Call transaction <tr.code> using <Bdctab>
Mode <A/N/E>
Update <S/A>
Messages <BDCMSG>.
Perform check.
Refresh BDCTAB.
Endloop.
7 Form check.
IF sy-subrc <> 0. (Call transaction returns the sy-subrc if updating is not successful).
Call function Format_message.
(This function is called to store the message given by system and to display it along with record)
Append itab2.
Display the record and message.
DIRECT INPUT
Thanks &regards,
Sravani

Your order is put on hold with order number 17772 issue in iStore

Hi All,
I am facing the following error in Oracle iStore R12 when changing the following profile options,
IBE: Perform Payment authorization in iStore -- "*Yes*"
ASO : Credit Card Authorization -- "*NO*"
And the error is
Your order is put on "Hold" with order number 17772.
Please help me on this how to i resolve the order put in Hold issue.
When trying to ship this order APP-WSH-231033: Pick release online has ended with warning. Please Check the Shipping Exception Report for further details.
In Exeception report: order 17772 for Picking as it has been placed on hold.
Could you please give me any Fix for this issue as soon as possible.
We are not using the Risk Management in Payee
Thanks
Vivek
Edited by: 912882 on 29 Sep, 2012 5:53 PM

hi,
please check this link it will help ful.
Credit card authorization for this order has failed

Unable to make a new call when she has an incoming call on hold

Cisco ip phone 6945 with dual-line dn configuration
when puting an incoming call on hold and trying to make a new call, she can not succeed.
Any suggestion to make it work,thanks

Pretty sure you are going to need an additional DN to make another call while having one on hold. Even though the calls don't appear to go over the second DN, you have to have as many "Lines" on the phone as calls active. the call on-hold, is still considered an active call on that device.

How can I automate the multi-step process of looking up a caller's contact details when a telephone call comes in with caller ID?

How can I automate the multi-step process of looking up a caller's contact details when a telephone call comes in with caller ID. I want to be able to see more than just the first & last name when a client calls me. I would like the caller ID to pull up the same information that shows up when you search for and find a contact name and then click on the right arrow.
My workaround is to force/enter both the last name and the address in the last name field since then the address will show up.
The only computerized way to look up this additional information is to put the client on hold, memorize the first and last name and correctly enter it in the search field of the contact database and hopefully find it before the client gets tired of waiting.
Note that I go to about 48 small jobs per week. So far I have about 9,000 contacts. It's easy to make a mistake and have the client wondering what I'm doing and what's taking so long.
The second workaround is to use paper and pencil and ask the client for name, address, cross street and phone number again, repeating what I did some time previously.
Please help meautomate this process.

You don't have to put anyone on hold. When the call comes in, tap the Home button, then tap contacts, go to the contact. Tap Home button when finished, tap the top banner to return to the call screen.

Clicking on PSTN calls only when placing call on hold/taking off hold

Hey all,
We have an ongoing issue that we thought was isolated to a call flow going to our UCCX, but after extensive troubleshooting, have found that it is occuring when a PSTN caller is being placed on hold by or being taken off of hold (most noticeable when taking off of hold) either via an IP Phone or another system (Unity Connection when releasing to switch is transferring outside of Unity Connection, UCCX and it's hold/unhold steps). If we mirror this call flow, only calling from an IP Phone to Another and being placed on hold, or including Unity Connection or UCCX in the call flow, this doesn't occur. We've been working with TAC and our ITSP for the past couple weeks but haven't really made much traction. Wireshark traces on the CUBE, so far, only have the audible noise on the ITSP facing interface of the CUBE, which is what I would expect given that it's not heard internally.
Call flow:
PSTN Caller -> ITSP via SIP -> CUBE -> SIP -> CUCM -> IP Phone
CUCM Version 8.5.1.15900-4
CUBE Version 15.1(3)T2
MOH and CUBE have a G711 region relationship. Most IP Phones and systems have a G729 relationship, but as a test I put a phone in my G711 test region and the issue still occurs.
The traces show the call being setup with G729 (preference on my dial-peer), but the stream is all G711 (audio from my IP Phone and MOH). This was verified both in the trace and via the CUBE when doing a "show sip calls <my cell phone number>" that both legs were G711 and the issue was still occuring.
We regularly restart the IPVMS service as our MOH changes from time to time, but again this is only happening on PSTN calls, so my focus at this time has been the CUBE.
Has anyone run into this before? Perhaps it's a bug and I need to update my CUBE's firmware?

Hi,
Did you enable “Play music on hold”?
If yes, you can try to disable it and then test again.
Please also check if there are any error message on FE Server when the issue happen.
Please also update to the latest version for Lync Server 2013 and have a test.
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support

Place Call and continue with remaining script steps - is it possible?

We recently picked up IPCelerate which has the ability to send a Text display or Text-to-Speech message to a group of phones simply by dialing a configured directory number. We would like to text our call center agents with this application when a caller has been waiting for X number of loops.
Attempting to use 'Place Call' to initiate this text page works, however, the script terminates immediately after the call is placed and we would like to continue on with the remainder of our script. In this case, on Success go back to the top of the loop.
I have also tried to use Call Subflow instead, thinking an executed sub-routine placing the call would return to the original script and continue on, but it didn't. In both cases, the system comes back and states " I'm sorry, we're currently experiencing system problems and are unable to process your call. Please try again later."
If there's a way to get this to work via Place Call or another process to accomplish the same result, that would be fantastic. I appreciate any feedback.
We're using UCCx Express 5.0 (2.064) with premium type licensing.

Hello,
I did get to an answer similar to what r.stockton did to solve the issue, but I left Exception (ContactInactiveException) Goto in ours to isolate exceptions because we have multiple contacts within our script. Turns out that this particular exception was thrown by the PlaceCall step ending abruptly by our paging system answering and hanging up the call quickly. MIVR logs showed that it didn't bomb as immediately as when the debugs showed it crashing - it actually went into a loop over and over with the Trigger contact before failing.
We had some help from TAC on this, but below is what we did to keep some error checking and notification we use available to us. We are running a check to make sure the Call Trigger (original call) is still active.
placecallContact = Place Call (to Number)
on Exception (ContactInactiveException) Goto
label:
CallActive = Get Contact Info (--Trigering Contact--, Active)
If (CallActive) Then
- True branch continues with script
- False Branch terminates call
If this takes the false branch, it means the trigger contact is not active, and has abandoned the call or been transferred elsewhere. Placed an End step here to prevent any loops.
If this takes the true branch, it means the triggering contact is actually still active. In our scenario, CallActive (boolean variable) is verified true and continues with the script without going into the loop of death.
I hope this helps!

Pick Up Call on Hold with Pickup Softkey

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