CCX using Place Call to hunt group. Unable to determine who answered

I'm trying to use the CCX to place a call to a CUCM hunt group and then once the answered, determine which member of the hunt group answered the call.
I have used several of the Get Info steps, including:
Get Call Contact Info
Get Enterprise Call Info
Get Contact Info
However, all of those steps always just report back the CUCM hunt pilot number, instead of the actual line group member who answered the call. Can anyone think of a way to determine the extension of the line group member that answered the call ?
If it matters, I am using CCX 9.0.2su1

I lack a system to test this on at the moment but perhaps try this CUCM Service Parameter. The question is whether CTI Manager will follow this logic or if this is only respected by the CDR flat file export.
Show Line Group Member DN in finalCalledPartyNumber CDR Field
This parameter determines whether the the finalCalledPartyNumber field in call detail records (CDR) shows the directory number (DN) of the line group member who answered the call or the hunt pilot DN. Valid values specify True (the finalCalledPartyNumber in CDRs will show the DN of the phone that answered the call) or False (the finalCalledPartyNumber in CDRs will show the hunt pilot DN). This parameter applies only to basic calls routed through a hunt list without feature interaction such as transfer, conference, call park, and so on. If a feature is involved in the call, then the hunt pilot DN will show in the finalCalledPartyNumber field regardless of the setting in this parameter. This parameter does not apply to Cisco CallManager Attendant Console.
This is a required field.
Default: False

Similar Messages

CUPC Incoming Call from Hunt Group

Hi all!
Is is possible that I can see whenever someone calls me on my extension, or whenever a call comes in from a hunt group? Can I differ the incoming calls?
Regards
Rene

Are you talking about the attendant console hunt group? If yes, then you can do this by using the broadcast hunting feature. What it does is, all calls to the hunt pilot will be placed in queue and will be displayed in the Broadcast calls window within the Attendant Console. All uses can see the calling number of calls in queue and pick and chose the calls they want to answer.
Check this link for more details
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/4_1/sys_ad/4_1_3/ccmfeat/fsccmac.htm#wp1144539
Regards,
Anup

CallerID not correct when calling a Hunt Group #

Not sure what happened, but when users called the 4357(Hunt Group) the caller id would show "Helpdesk and also the callers name and ext" on the caller ID of the users in the 4357 (Hunt Group List).
Now when users call the 4357 the caller id on the (Hunt Group) end just shows the "callers name and ext".
We like when it would show Helpdesk along with the user info because we know the person is actually calling the Helpdesk and not calling directly. For example, my boss knows to only pick up calls that come into the helpdesk when he is the only one here, but now he is picking them all up because it looks as though they are calling him directly.
Any suggestions?

Hi Aaron,
I have made the changes you told o make and its working now.
Thanks.
Syed Kazi.

Extending the Hunting time on a Hunt Group CUCM 8.5.1

Good afternoon. We are trying to implement a hunt group for our main switchboard. The scenario would be all calls to the home office come into a hunt pilot and then hunt, top down, to the main switchboard extension and then to a choice of 3 other devices that would be logged in. The problem is that we cannot get the hunt group to hunt longer than 1 minute. I have changed the RNA Reversion Timeout and with all the devices logged it, the call will only hunt until the 60 seconds has expired and then the call drops; 15 seconds at the RNA Reversion Timeout will get to all 4 devices - anything longer and the call drops before finishing the hunting process. From the information that I can find, a hunt group should "hunt" for up to 30 minutes (or so we read in the Hunt Pilot information). We're using a "Top Down" algorithm since we want all calls to always start at the switchboard directory number, and all of the Hunt Options are set to "Try Next member; then, try next group in Hunt List". The only place where we have indicated any time is on the Line Group - RNA Reversion Timeout. There are times where only the switchboard device may be logged into the hunt and we need to have that ring time at 60 seconds minimum for the call to be handled appropriately.
Is this a simple overlook on our part or are we limited in what we want to do?
Thanks for any insight that you can give us.
Cinde

Hi Cinde,
I just wanted to add a note to the excellent tips from my friend Aaron here (+5 "A")
The Maximum Hunt Timer is only leveraged when considering it in conjunction with the Call Forward settings on the Hunt Pilot config as noted here;
Maximum Hunt Timer
Enter a value (in seconds) that specifies the maximum time for hunting. Valid values specify 1 to 3600. The default value specifies 1800 seconds (30 minutes).
This timer cancels if either a hunt member answers the call or if the hunt list gets exhausted before the timer expires. If you do not specify a value for this timer, hunting continues until a hunt member answers or hunting exhausts. If neither event takes place, hunting continues for 30 minutes, after which the call gets taken for final treatment.
Note If hunting exceeds the number of hops that the Forward Maximum Hop Count service parameter specifies, hunting expires before the 30-minute maximum hunt timer value, and the caller receives a reorder tone.
In addition, Cisco Unified CM only uses the configuration for the Maximum Hunt Timer setting if you configure the Hunt Forward settings in the Hunt Pilot Configuration window.
From;
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/8_5_1/ccmcfg/bccm-851-cm/b03htpil.html
There is an old bug that describes this in better detail;
Maximum Hunt Timer (MHT) description is unclear
CSCsg93573
Description
Symptom:
In the Hunt Pilot form, under Hunt Forward Settings, there is a field
called Maximum Hunt Timer (MHT). The description in the online help is misleading.
Currently, it conveys that the setting will be used when calling the Hunt Pilot
directly, which is not true. This setting is only used if Hunt Forward Settings
are configured.
So, to accomplish what you want you'll need to set up and link multiple Line Groups that contain the same 4 member DN's via the "Try next member,then try next Group in Hunt List" setting.
Let's say in our example the members of the first Line Group Called Main#1 are 3333,4444,5555 and 6666 then you create a second Line group called Main#2 that contains 3333,4444,5555 and 6666 and Main#3 that contains 3333,4444,5555 and 6666.
Include Main#1,Main#2 and Main#3 in the Hunt List; and Switch the rule on the first Line Group
to, Try next Member;then try next Group in Hunt List and Switch the rule on the Second Line
Group to, Try next Member;then try next Group in Hunt List.
This will then cycle through the Members 3333,4444,5555 and 6666 three complete times.
Cheers!
Rob

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!

How to enable HLog for B-ACD Hunt Group

We are using B-ACD for imcoming calls, the hunt group for BACD has 5 members. I want people to be able to login/logoff from this hunt group.
I have this working with the DND button, but then the phone doesn't ring when the extension is dailed directly. So it seems the alternative is using another feature/button called "Hlog".
I've added the HLog softkey but nothing happens when its pressed. I can't seem to find instructions on what else might need to be done to enable HLog.
Here is a bit of the related config info
ephone-template 16
softkeys idle Redial HLog Cfwdall Pickup Dnd Mobility
softkeys seized Cfwdall Endcall Redial Pickup Gpickup Callback
softkeys connected Trnsfer Endcall LiveRcd Hold Mobility
ephone-dn 15 dual-line
number 206
no-reg primary
label 206
description Kathy
name Kathy
call-forward busy 299
call-forward noan 299
timeout 10
ephone 6
device-security-mode none
video
mac-address 0021.1BFC.BF2B
ephone-template 16
max-calls-per-button 2
username "kathy"
password 1112
type 524G
button 1:15
ephone-hunt 50 sequential
pilot A888
list 204, 205, 201, 206
timeout 8, 8, 8, 8
no-reg pilot
statistics collect
auto logout 3
description *** CCA BACD ***
display-logout All agents logged out

We still don't have Logging in and out of the BACD Hunt group working.
The #350 and *350 don't seem to work. On the phone when I dial them it connects for a second then disconnects, no feedback if it worked. But when I try calling our business number the phone still gets rung in the hunt group sequence. So the phone isn't getting logged out.
On some of the phones I can see the "HLog" button but when pressed it says "that key is not active here".

Hunt group via JTAPI

When someone calls a hunt group, and call gets transfered over to an agent, we need to know what was the original number that the caller dialed. I can get the extension or MAC of the phone receiving the call and the caller ID, but not the original number called. Is there a way to look that up via JTAPI or AXL or a combination of both?
Thanks

not sure about JTAPI or AXL, but if you don't need it immediately, you could get the original calledpartynumber from the CDR of the first leg of the call ...

7961GE Hunt Group ringing name

Our shop has been exclusively 7961Gs for 7 years now. We recently purchased some 7961GEs and are noticing that when a phone is part of a Hunt Group, the Hunt Group no longer displays on the inbound call. On the 7961Gs it would say the From: <Caller> For: <Hunt Group> .
Our Help Desk personnel love the 1Gbit speed, but hate not knowing if the incoming call is for them or the Help Desk.
Any help is much appreciated!!
Thank you,
--Whammer

Hi Whammer,
I'm guessing the new 7961GE's are running a pre - 9.0 firmware;
It sounds like this bug
CSCth95017 - Configurable parameter on how callingparty is displayed for hunt groups.
Description
Symptom:
As of phone load 9.x and higher, calls to a hunt pilot show the following format for the calling party on the ringing phone.
From 'Calling Party Name' for 'hunt pilot DN'
i.e. From John Doe for XXXX
The previous format was
From 'Calling Party Name' ('Calling Party Number')
i.e. From John Doe (XXX-XXX-XXXX)
Need a configurable parameter to change the format to the previous format if the customer desires.
Conditions:
Occurs on phone loads 9.0(x) and above.
Workaround:
Downgrade to 8.4.x
Details
1st Found-in:                          (1)
8.6(2)
More
Less
Status:
Fixed
Last Modified:
Mar 11,2012
Known Affected Versions:
More
Less
Fixed-in:                          (3)
9.0(0.98000.74), 9.0(0.98000.21), 8.6(2.21023.1)
More
Less
Product:
Cisco Unified Communications Manager (CallManager)
Platform:
Dependent
Severity:
2 - severe
Customer Reported:                          (26)
Cheers!
Rob

Setting up Hunt Group using an extension number instead of a pilot number

Hi Guys,
I would like to set up a hunt group. My question is can I use a real extension number assigned to a phone to be a pilot number instead of creating a new pilot number.
The requirements are if no one is answering a call to an extension 1000 (extension 1000 is assigned to a user) it should be forwarded to 1001, 1002, 1003 and back to 1000. I was thinking of using 1000 as pilot number. Is that possible ?
Thanks Guys

Hi Guys,
Thank you very much for your suggestions. Here is what I have done:
With Manish's solution the call forwarding works upto the third phone then it got disengaged. Looks like there is a timer I need to increase so that the call will not time out. Could you please let me know which timer I have to adjust. I have checked the max forwards hops and they are set to 15-20 so don't think this is an issue.
With regard to Jamie's solution, I have created a separate partition and assigned same DN 1000 to it. It works great when I call from inside. However when I call it from outside using the full E164 number the call goes to 1000 then it just go to the voicemail of 1000 if not answered. I have removed the settings under DN page so that it will not go to voicemail. After doing this the phone with extension 1000 keeps ringing until it times out. Not sure where the issue is here.
I have given a suggestion to my customer to use a new pilot number with hunt groups. I have tested this as well. When I call it from outside the call goes to all the five phones and it just keep going into circle until someone answered and that's what I want. But when I call DN 1000 from inside it goes to all the five phones and when I reaches to the last phone configured under the line group member the call then just dropped out. Is there a way to fix this so that when someone just dial the extension number from inside all the phones keep ringing in circle manner and did not drop until someone has answered it.
Thanks a lot guys.
Regards

Hunt Group with Pots lines in Call Manager

Hello all. I have a site with 12 Pots lines. It is all configured via Call Manager. We are using mgcp. I asked the telco to setup the numbers to hunt which they said they did but still when I call the main number second time it just rings and eventually get some voicemail that is not from the call manager. I think I need to setup Hunting in the Call Manager? Is that correct?
Line Group --> Add the extensions in here and set them up to hunt?
Hunt List --> Create a Hunt list and associate a Line Group with it?
Hunt Pilot --> I add all the extensions in here?

Let's take one step back. What are you trying to accomplish? I am assuming you want to market one number but have the ability to accept multiple calls on that number. If that is the case, having the telco set up a hunt is the way to go.
Beyond that, where do you want those calls from the PSTN to route? Are you wanting those calls to ring a hunt group on your side? If so, you will need to do the following:
     1. Configure a Line group. (Call Routing -> Route/Hunt -> Line Group) This is where you will be adding the extensions. You will also need to specify a distribution algorithm. The distribution algorith, is where you will be choosing how you want the extensions to ring. For instance, if you set the DA to broadcast, every extension in the line group will ring at the same time. Click Help --> this page, while you are in the line group configuration page to get more information about the settings you can modify and different algorithms you can use.
     2. Configure a Hunt List. (Call Routing -> Route/Hunt -> Hunt List) You will need to create a new Hunt List and add your line group to it. The hunt list is evaluated in a top down fashion. You can get pretty tricky with multiple line groups with different distribution algorithms in your hunt list, but if you just want something basic,just add the line group you made.
     3. Configure the Hunt Pilot. (Call Routing -> Route/Hunt -> Hunt Pilot) The hunt pilot will be the dialed number that triggers the hunt group. Make sure you select all the required settings. You will be selecting the hunt list you just made in this configuration page. Be sure to set a maximum hunt timer and final destination so that the calls do not just ring forever. You can get pretty granular with the settings in here. Reference the help --> this page document for more information.
     4. POINT THE FXO PORTS TO THE HUNT PILOT. This is very important. Calls to your analog lines will not trigger the hunt group unless you point the ports to the Hunt Pilot. If you are using MGCP/SCCP you will need to enter the hunt pilot in the "Attendant DN" field for EVERY FXO port. If you do not enter a value here, CUCM will have no idea what to do when the port rings. It will not answer the call if there is no value specified. This would explain why your calls were ending up with the telco voicemail service; CUCM was never answering the call.
If you completed the above steps correctly, you should be able to call the main number multiple times and have the calls go to a hunt group on your side, no matter which line the call ends up coming in on.
Let me know if you have any questions, this can be kind of tricky at first.
PS, if the calls were going to a voicemail that wasn't from Call Manager (Unity Connection) call the telco and have them remove voicemail service from the lines, assuming you have configured Unity Connection to handle that.
Hope this helps,
Dallan

Call Pickup within a Hunt Group CallManager 5.1

I have a customer where we have several hunt groups configured. We also have Pickup Groups configured for each of these areas so that anyone within that group of users can pickup a call ringing on another phone. If someone dials a number directly that is within the pickup group, the call can be picked up from another phone in that group. However, if someone dials the hunt group number, other people within the pickup group cannot pickup the call and have to wait until the hunt works its way around to them. The DNs in the Pickup Group are the same as the DNs in the Hunt Group.
What I want to know is "should this work?". I think I understand why it isn't working, as the dialled number is not a number within the pickup group, as it's a hunt pilot, but is there any way of getting this to work?
Any help would be appreciated.
regards
Mark.

Hi Mark,
You are hitting this bug CSCsb12946 which prevents this feature from working. For various CCM versions there was this workaround but supposedly not for CCM 5.1 (see Fixed-In versions);
Set enterprise parameter (System->Enterprise Parameters->Cisco Support Use 1) Cisco Support Use 1 to CSCsb42763 (case sensitive)
This operation added listbox (Route Plan->Route/Hunt->Hunt Pilot->hunt pilot XXXX -> Call Pickup Group) Call Pickup Group.
From this doc;
http://www.ciscotaccc.com/kaidara-advisor/voice/showcase?case=K11612191
1.- Enter
CSCsb42763 (in mixed case) in the Enterprise Parameter 'Cisco Support
Use 1' under the new group 'Cisco Support use Only' and click update. Only the value
CSCsb42763 in this parameter can enable the configuration. Change the
value to anything else to disable this functionality.
2. Go to Hunt Pilot Configuration page and select a call pickup group from the Call
Pickup Group dropdown list and click update.
CSCsb42763
Call Pickup fails on Hunt List
Issue:
Call pickup fails for hunt list when call is CFA from Route Point in 4.1(2)ES41. This is not the designed functionality for this issue and will break the Call Pickup logic.
Workaround:
Use multiple shared lines & CFNA between each shared line.
Related Bugs
Call Pickup fails when ringing line list member
Symptom: Selecting PickUp or GPickUp does not pickup the call from any of the members of a line group. Condition: CM 4.x Workaround: None. Feature not supported. Further Problem Description: Defect CSCsa66224 may appear to have allowed this feature to work in earlier versions.
Call Pickup from line group member fails after es installed
Symptom: A call is CFNA to a hunt pilot/line group. The call can no longer be picked up. Conditions: Calls CFNA to a hunt pilot. 4.1(2)ES36. Workaround: None
1st Found-In
4.1(3)
4.1(2)
Fixed-In
4.2(0.35)
4.2(0.841)
4.2(1.15)
4.1(3)SR2
4.1(3)ES30
4.1(3)ES29
4.1(3)ES20
4.1(2)ES43.1
5.0(2.1000.3)
5.0(2.50.17)
5.0(1.99.33)
5.0(1.50.224)
5.0(1.20.84)
5.0(0.319)
4.2(1.2)
4.2(0.750)
4.2(0.24)
I did se this one Post from John regarding CCM 5.1.X,
http://forum.cisco.com/eforum/servlet/NetProf?page=netprof&forum=Unified%20Communications%20and%20Video&topic=IP%20Telephony&CommCmd=MB%3Fcmd%3Dpass_through%26location%3Doutline%40%5E1%40%40.1dddea7c/10#selected_message
Hope this helps!
Rob

CUCM 8.6 - Call Forward All Doesn't Work With DN In Hunt Group

Hello,
I'm trying to make a hunt group that's sole purpose is to dial out to our Help Desk member's cell phone (on-call after hours). The idea is that we want the hunt group to broadcast the call to the on-call engineers. Whoever picks up first takes the call.
At least that's the idea now. They may want to change that to - ring primary on-call cell phone, if no answer ring secondary on-call engineer.
The target cell phones will change every week and I wanted to give the supervisor a means of changing the numbers through the ccmuser page.
I created two DNs and placed them in a hunt group, each with call forward all to a cell phone number.
It seems that the hunt group ignores the call forward all setting and that is by design (https://supportforums.cisco.com/document/9126/cisco-callmanager-call-forward-all-cfa-does-not-work-if-line-hunt-group).
Does anyone else have any suggestions on how to accomplish this?

That's correct, what you're seeing is WAD, this topic has been discussed plenty of times before, the only option you have is SNR

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.

CCME Call Forward from one Hunt Group to another Hunt Group Failure

Hi I have a couple of hungroups in Cisco Call Manager Express. I am trying to configure a Call Forward no answer from one hunt group to another. Does anybody know if this is possible? If so, is there a config available? Here is my config, but it is not working.
Thanks,
Derek
voice hunt-group 20 parallel
final 2290
list 2201,2251,2252
timeout 30
pilot 2209

Hi Derek,
This is supported with the restriction shown below;
final number
Router(config-voice-hunt-group)# final 8888
Defines the last extension in a voice hunt group.
If a final number in one hunt group is configured as a pilot number of another hunt group, the pilot number of the first hunt group cannot be configured as a final number in any other hunt group.
From;
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeadm/cmecover.html#72373
Cheers!
Rob

Lync RSG Hunt Group not working as expected, not getting calls on lync clients

I have created an Hunt Group in Lync. When i dial the number i do get the welcome message but after that it does not get forwarded to the test user which is configured in the Group. (Created a Queue and selected the group here and assigned this to the workflow)
Error on Lync report:
Request: BYE Response: 200
Diagnostic ID: 26029 Category: Unexpected
Component: Response Group application
Reason:
Call is disconnected because of invalid configuration.
Description:
Call is disconnected because of invalid configuration.
Error in Event Viewer:
The workflow runtime encountered a critical error.
Failure occurrences: 8, since 4/17/2012 7:47:14 AM.
The last encountered error was from a workflow with the display name: TestRG, the URI: sip:[email protected], and the GUID: 810c628c-d0c5-4b31-b8d5-f61595def5b0
Suhas

HI,
Do you deploy standard lync server or enterprise lync server?
If enterprise lync server, please log in SQL server and search the rgsdyn--dbo.ActiveFEInstance. Please check you can find the FE server FQDN and the port 8404 in this table.
Please running the Lync server logging tool on the FE server to find why the Response Group can not transfer the calls to the agent.
Please remember to click “Mark as Answer” on the post that helps you, and to click “Unmark as Answer” if a marked post does not actually answer your question. This can be beneficial to other community members reading the thread.

CCX using Place Call to hunt group. Unable to determine who answered

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