CUCM to CVP calls. CTI-RP vs Route Pattern
CVP 9 or above
CUCM 9 or above
Requirement:
1. Consultive Warm Transfer - The agents to be able to transfer calls to a a different department by dialing an internal number and wait in the queue until answered.
2. Internal - Back-office people to dial internal IT-Helpdesk or HR
I see the above call flows as same, i.e. a Call Originating from CUCM to CVP .... correct me please?
I have tested both and they both work exactly the same way, i.e. using a CTI-RP associated to PGUSER, ICM answers it sends correlation id to CUCM and CUCM sends this to CVP ...AND... using a simple route patters instead point to CUCM-CVP SIP trunk. Functionally they behave same way - ICM/CVP answers and queues call until answered.
But the documentation confuses me, below snippet from CVP Config Guide
"... Calls Originated by Unified CM
Internal Help Desk calls: For these calls, the Unified Communication Manager (CM) phone user calls a CTI Route Point
Consultative Warm Transfer: For these calls, a Unified CM agent places the caller on hold and dials in to Unified ICME to reach a second agent .... "
And then on the same doc, there a Note
(*1) Note For warm transfers, the call from Agent 1 to Agent 2 does not typically use a SIP Trunk, but you must configure the CTI Route Point for that dialed number on the Unified CM server and associate that number with your peripheral gateway user (PGUSER)
(*2) And then again on the same doc under 'Unified ICME Warm Consult Transfer/Conference to Unified CVP' chapter/section it mentiones doing this using a Route Patter 'Create a route pattern and assign the route list to the route pattern'
So the confusion is
1. Why treat these call flows as Internal and Warm Transfer - they are calls from CUCM to CVP for the same end result - queue the call and transfer to an agent?
2. Route pattern or CTI-RP, what diff it makes? They both behave the same way, so is there a diff from reporting point of view that a call to CTI-RP are treated as Transferred rather than new calls or what?
3. Also if you compare (*1) & (*2) above, they both talk about Warm Transfer and *1 says 'must use CTI-RP' and *2 says use a Route Pattern?
Please assist.
Thanks & Regards,
Kartik
Kartik,
The Route Pattern that is mentioned is used for connecting a call leg through CVP to a local VXML Gateway for media playback. The CTI Route Point is entirely different from the Route Pattern/Route List setup. Here's the basic call flow:
Internal caller dials DN
DN hits CTI RP in CUCM. CTI RP sends call to ICM.
ICM matches DN to Call Type to Script, executes Script.
At some point, Script has either Send To VRU or a Run Ext. Script node.
ICM sends CUCM Network VRU label back to CUCM.
CUCM routes label using Route Pattern and Route List. The CSS of the internal caller determines how this is modified, i.e. which prefix digits to add for determining VXML gateway to route to.
Call is sent to CVP through SIP trunk
CVP receives call, tells ICM it has the call.
CVP starts new call leg to VXML gateway with digit string to match bootstrap dial-peer.
VXML Gateway receives call, initiates bootstrap TCL and VXML magic.
Yes, this is basically the same call flow for a fresh internal call to a queue, or an internal warm transfer to a queue. The CTI Route Point is needed in both cases. The Route Pattern/Route List combo is needed in both cases.
When you start looking at reporting, yes of course the two call scenarios are different. One is a transfer, the other isn't. The transferred call will have a more complex call history if you look at it in the TCDR.
From the standpoint of call legs, you will use less legs if you do a direct (one-step) transfer instead of a warm transfer. It is also simpler to maintain the call context in that case. In a warm transfer scenario, the agent is putting a caller on hold, then starting a new call, and joining the two calls together. The new call is coming from the agent, not the original caller. In a direct transfer, CVP just takes back the original call, potentially does more queuing, then sends the original caller to a new agent target.
-Jameson
Similar Messages
-
Route pattern to SIP trunk problem
Hello, I have a 2801 router that has been configured with CME and a working SIP connection to my local ISP.
Tested with calls via CME so I know for sure that the SIP config and dial plan is fine on this gateway.
Next I wanted to try out CUCM so I set up a CUCM 8.6 box that is connected to the 2801 router to use as it's SIP gateway.
The only change I made to the gateway router config was to alter the "ip option 150" address so that the phones go to CUCM for their configs etc (which they do with no problems).
Then I set up a SIP trunk in CUCM along with a route pattern which is to use the SIP trunk within the Gateway/Route list option.
But when I make a call that matches this route pattern all I get is the intermittent beep message from the phone. I cannot route calls succesfully through it.
I have checked network connectivity and all is fine. The IP address I specfied in CUCM for the SIP trunk is simply one of the interfaces on the 2801 router and it is definitley reachable.
I also activated "debug ccsip all" on the 2801 gateway router but nothing appears. So it seems like the calls are not even reaching the 2801 gateway ?
Is the problem possibly a conflit between CME on the gateway router and my CUCM ?
Do I need to disable CME somehow on the gateway first ? Or am I not doing something correct in the CUCM config ?
Thank you kindly for any suggestions.
ps. I have attached a couple of screenshots of my config.Hello, thanks for helping.
I activated "debug voice ccapi inout" as well as "debug ccsip all" on the gateway but nothing showed up.
Therefore I deduce the call is not even making it to across the SIP trunk into the gateway router ?
As I am a newbie trying this out for the first time, it is guranteed to be something really simple.
I have included my running config from the gateway router below..
One addition I made was to add an incoming dial peer. That is "dial peer 5, description CUCM SIP trunk".
I set it up with a destination patter 2... to match my phone config on CUCM which have numbering in the 2000 range.
Sorry, I got RTMT up and running but could not get any meaningful results from it. I need to learn up on that.
I did however run a 'dialed number analysis' from CUCM direct and have attached the result. It seems the dialled number "99" is matching the route pattern OK.
So why is it not then moving down the SIP trunk to my gateway and getting picked up by the incoming dial peer ?
Thanks if you guys can offer any more help.
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router
boot-start-marker
boot system flash:c2801-ipvoicek9-mz.151-2.T0a.bin
boot-end-marker
no aaa new-model
clock timezone nzst 13 0
dot11 syslog
ip source-route
ip dhcp pool DATA_SCOPE
network 192.168.200.0 255.255.255.0
default-router 192.168.200.1
dns-server 8.8.8.8
ip dhcp pool VOICE_SCOPE
network 192.168.100.0 255.255.255.0
default-router 192.168.100.1
option 150 ip 192.168.2.115
ip dhcp pool MGMT_SCOPE
network 192.168.1.0 255.255.255.0
default-router 192.168.1.99
ip cef
ip name-server 4.2.2.2
no ipv6 cef
multilink bundle-name authenticated
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g729r8
codec preference 3 g711ulaw
codec preference 4 ilbc
voice translation-rule 1
rule 1 /^9/ //
voice translation-profile Strip9ToGetOut
translate called 1
voice-card 0
crypto pki token default removal timeout 0
crypto pki trustpoint TP-self-signed-2995340181
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-2995340181
revocation-check none
crypto pki certificate chain TP-self-signed-2995340181
certificate self-signed 01
3082023E 308201A7 A0030201 02020101 300D0609 2A864886 F70D0101 04050030
31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
69666963 6174652D 32393935 33343031 3831301E 170D3733 30363034 31393534
32305A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649
4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D32 39393533
34303138 3130819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281
8100C34D C8ECBB53 E01373A3 2E286B78 2D23042B 1C8588B1 A7861899 BA1C6860
AE1D7868 2A59E3BC 54D0A457 8FFDE27F C09104E5 C7A429F3 74CD9DA8 4A980366
675CC27C CDB94838 821CC05F 2C0AC2BC D882C132 6CAA1FA6 6DA740E4 562428B1
12B741F1 A50C9246 4CC35EDA DEE1D038 3883BB35 A91ABF8B 483E4160 F5FA4B5A
9A570203 010001A3 66306430 0F060355 1D130101 FF040530 030101FF 30110603
551D1104 0A300882 06526F75 74657230 1F060355 1D230418 30168014 72119640
F3396E1F E4168086 D31D8619 0D8337FF 301D0603 551D0E04 16041472 119640F3
396E1FE4 168086D3 1D86190D 8337FF30 0D06092A 864886F7 0D010104 05000381
81003B5A 29DE3A1E C5AB6092 E8D90650 C80752FC 0AAC93FD C5DE3D69 071B08FA
D4013232 81CA07E7 15F90190 6A3AD6A0 1D05F0F2 13479568 888332A5 F81E2681
7DA44095 4D11CFB7 CA79579A 8D95DE54 7B00173C E2C50573 A310C8C9 1487FEFC
CE35B66E 9EF94CFA 8D6D6DCD ADC78132 2709F198 6DF2F0FA D80CC088 D0C4C7D1 080B
quit
license udi pid CISCO2801 sn FTX0947W07M
username xxx privilege 15 password 0 xxx
interface FastEthernet0/0
ip address 192.168.3.50 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
no ip address
duplex auto
speed auto
interface FastEthernet0/1.2
encapsulation dot1Q 2
ip address 192.168.2.1 255.255.255.0
interface FastEthernet0/1.99
encapsulation dot1Q 99
ip address 192.168.1.99 255.255.255.0
interface FastEthernet0/1.100
description voice_VLAN
encapsulation dot1Q 100
ip address 192.168.100.1 255.255.255.0
interface FastEthernet0/1.200
description data_VLAN
encapsulation dot1Q 200
ip address 192.168.200.1 255.255.255.0
ip forward-protocol nd
ip http server
ip http authentication local
ip http secure-server
ip route 0.0.0.0 0.0.0.0 192.168.3.1
logging esm config
tftp-server flash:/phone/7940-7960/P00307020200.bin alias P00307020200.bin
tftp-server flash:/phone/7940-7960/P00307020200.loads alias P00307020200.loads
tftp-server flash:/phone/7940-7960/P00307020200.sb2 alias P00307020200.sb2
tftp-server flash:/phone/7940-7960/P00307020200.sbn alias P00307020200.sbn
control-plane
mgcp fax t38 ecm
dial-peer voice 1 voip
description local_7_Digit_Calling
translation-profile outgoing Strip9ToGetOut
destination-pattern 9[2-9]......
session protocol sipv2
session target ipv4:203.184.16.2
voice-class codec 1
dial-peer voice 2 voip
description international_calling
translation-profile outgoing Strip9ToGetOut
destination-pattern 900T
session protocol sipv2
session target ipv4:203.184.16.2
voice-class codec 1
dial-peer voice 3 voip
description national_calling
translation-profile outgoing Strip9ToGetOut
destination-pattern 90[34679].......
session protocol sipv2
session target ipv4:203.184.16.2
voice-class codec 1
dial-peer voice 4 voip
translation-profile outgoing Strip9ToGetOut
destination-pattern 90[34679].......
dial-peer voice 5 voip
description CUCM SIP trunk
destination-pattern 2...
session protocol sipv2
session target ipv4:192.168.2.115
voice-class codec 1
sip-ua
authentication username xxxxxxxxxx password xxxxxxxx
060
telephony-service
max-ephones 10
max-dn 20
ip source-address 192.168.1.99 port 2000
load 7960-7940 P00307020200
max-conferences 4 gain -6
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 1 dual-line
number 1000
name Lydia Francis
ephone-dn 2 dual-line
number 1001
name Leah Francis
ephone-dn 3 dual-line
number 1002
n
ephone-dn 4 dual-line
number 1003
ephone 1
mac-address C80A.A970.01DE
type CIPC
button 2:2
ephone 2
mac-address 000C.3070.8705
button 1:1 2:15
ephone 3
mac-address 000C.8546.5954
button 1:3 2:15
line con 0
logging synchronous
line aux 0
line vty 0 4
privilege level 15
login local
transport input telnet ssh
scheduler allocate 20000 1000
ntp server 195.43.74.123
end -
Hi All;
I am gettting the below messages on the CVP Call Server version 8 and actually the Call Server is out of service, any advise?
The CVP PG is up and activated, the CVP Call Server registered on the Gatekeeper and I saw this on the gatekeeper, also the VXML Server is UP. But when I try to browse the statistics page using the Operational and Consol Server, then it also gives a message that not able to reach it, also its status at the operational and consol is down. Below are the messages I see it:
At CVP Call Server:
14:26:41 Trace: INFO: H323CallMgr::sendRAI: Successfully sent RAI for resource unavailability
14:27:26 Trace: INFO: H323CallMgr::sendRAI: Successfully sent RAI for resource unavailability
Unable to retrieve statistics for Unified CVP Call Server with IP Address: 10.180.22.137 and Hostname: vivadrcvp at this time.
14:26:41 Trace: INFO: H323CallMgr::sendRAI: Successfully sent RAI for resource unavailability
14:27:26 Trace: INFO: H323CallMgr::sendRAI: Successfully sent RAI for resource unavailability
At the Operational Console:
Unable to retrieve statistics for Unified CVP Call Server with IP Address: 10.180.22.137 and Hostname: vivadrcvp at this time.
What could be the reason? Is it license?
How can I know if the license if not valid?
Regards
BilalDear Geoff;
I am facing the same thing, but in the OAMP is shown to be down and not able to get any statistics on this CVP Call Server.
Actually the PG01, PG02 and PG03 are enabled in the router registry (while PG01 for CUCM PG, PG02 for CVP Call Server PG and PG03 for the Media Routing PG).
It was working before and I was receiving calls on it, but suddenly this happened.
Actually, an upgrade happened from version 7 to version 8 and we imported new licenses for the VXML Server but did not import new licenses for the CVP Call Server. Could be a license issue because we have to import the new license to change from version 7 to version 8?
Thanks in advance for the help.
Regards
Bilal -
Passing CLID and AgentID for CVP Call Transfer
Hi, all.
Currently we are using CVP 7.0(2) integrated with ICM 7.5(9), Just want to check if the following is feasible:
1. Caller calls the hotline being routed to a queue.
2. An agent picks up the call.
3. Agent does a consult transfer to another ICM routing script configured with another Dialed Number. It is a post call survey.
4. The post call survey takes over the call when the agent complete the call transfer
5. The caller's no (CLID) and AgentID are passed over to the ICM Routing script of this post call survey.
Can we achieve the above mentioned scenario particularly on item 5? Would appreciate any help and advice.
Thanks & Regards,
EricHi, Geoff.
Thanks for your reply. I have copied and pasted here a list of steps which you have posted previously about the CVP Warm Transfer.
1. PG Explorer - CUCM PIM Routing Client - Network Transfer Preferred not checked
2. PG Explorer - CVP PIM Routing Client - Network Transfer Preferred not checked
3. PG Explorer - CUCM PIM Advanced - Network VRU - NONE
4. PG Explorer - CVP PIM Advanced - Network VRU - Type 10
5. NVRU Explorer - Type 10 Network VRU, label for the CUCM routing client associated with the customer instance. Let's say this is 8222222222.
6. NVRU Explorer - Type 10 Network VRU, label for the CVP routing client associated with the customer instance. Let's say this is 8111111111.
7. Dialed Number List - dialed number for the incoming call associated with the customer instance. This dialed number is on the CVP PIM Routing Client. This DN is associated with a call type which is then mapped to the initial script.
8. Dialed Number List - transfer dialed number associated with the customer instance. This dialed number is on the CUCM PIM Routing Client. The transfer dialed number 3151 is associated with a call type which is mapped to the transfer script.
9. DNP. The number transferred to from CAD is 3141 which is a pattern in the DNP that maps to the Dialed Number 3151 with a post route to CUCM PIM Routing Client. The DNP Type is "PBX" - and PBX is set up in the Agent Desk Settings
10. Agent Desk Settings - All but "Operator" are checked
11. When the second call is placed for the warm transfer, the label defined on the CUCM RC plus the correlation ID will be sent back via EAPIM/JGW to CUCM (for example, if the label is 8222222222, with a correlation ID it could be 822222222212345) since the call originated from the CUCM RC and since the NetworkTransferPreferred check box is not checked.
12. A route pattern 8222222222! in CUCM sends the call down a SIP trunk to CUPS.
13. CUPS has a static route on 8222222222* to send the call to the CVP Call Server.
14. CUPS has a static route on 8111111111* to get the IP call to the gateway. Note that in a branch office deployment, TDM calls into the gateway use "Send to Originator" pattern in the Call Server to force the transfer label back to the voice gateway; so this pattern in CUPS is ONLY used by VoIP calls.
15. In all preliminary scripts that get the customer to the agent, set the variable Call.NetworkTransferEnabled to the value 1. This is set before the transfer is called.
18. For the device targets, you need a label on the CVP RC, but you do not need one on the CUCM RC, so do not add one.
I did try the above mentioned steps in my system, the transfer works alright. The only difference is that I did not set the variable Call.NetworkTransferEnabled to the value 1. To the ICM, is is considered as Internal Out Transfer or External Out? Another thing I have noticed is that when the call transfer takes place, the IP phone would display the Route Pattern of the CM Label + correlation ID configured, e.g. 822222222212345. this is rather confusing for the agents when they see this long string of number as it does not make sense to them. Is there any way to display some other meaningful line display or number (E.g. Transfer to mainline or 2900) while maintaining the CM label pattern 8222222222!?
Thanks in advanced.
Regards,
Eric -
CUCM 8.6 Call Forwarding to External Number Issue
Hello,
Call forwarding worked without problems, we could forward our phones to external numbers and everything was ok, when somebody called to my phone, I could got the call to my cell phone.
But now when I forward my phone to external number and try to call to my phone I get busy trigger.
We didn't change configuration or install any update.
I think its my ISP-s problem, to whom we have SIP Trunk.
I don't understand log file, so can you tell what is the problem?
Here is log:
057729XXXX is called party, cell phone number
original calling party number is 240XXXXX, but it is forwarded to 2484XXX
INVITE sip:2484XXX@ISP-IP:5060 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
From: <sip:057729XXXX@MY-CUCM>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Cisco-Guid: 0383266432-0000065536-0000191815-2219117834
Session-Expires: 1800
P-Asserted-Identity: <sip:057729XXXX@MY-CUCM-IP>
Remote-Party-ID: <sip:057729XXXX@MY-CUCM-IP>;party=calling;screen=yes;privacy=off
Contact: <sip:057729XXXX@MY-CUCM-IP:5060>
Max-Forwards: 68
Content-Type: application/sdp
Content-Length: 215
v=0
o=CiscoSystemsCCM-SIP 4052091 1 IN IP4 MY-CUCM-IP
s=SIP Call
c=IN IP4 MY-CUCM-IP
t=0 0
m=audio 29790 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
|2,100,56,1.173711429^MY-CUCM-IP^MTP_3
17:34:18.526 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 2|2,100,56,1.173711429^MY-CUCM-IP^MTP_3
17:34:18.526 |EnvProcessUdpHandler::fireSignal - SEND: index = 2, handler = 0xb2d59c98|*^*^*
17:34:18.526 |EnvProcessUdpPort::fireSignal - SEND, destination = ISP-IP:5060|*^*^*
17:34:18.526 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 1172, ISP-IP:5060)|*^*^*
17:34:18.536 |EnvProcessUdpHandler::handle_input - handle = 334|*^*^*
17:34:18.536 |EnvProcessUdpHandler::handle_input Status: 0, Id: 2|*^*^*
17:34:18.536 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 358 from ISP-IP:[5060]:
[12623361,NET]
SIP/2.0 100 Trying
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
CSeq: 101 INVITE
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
Server: CISCO-SBC/2.x
Content-Length: 0
|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0x0/ccsip_spi_get_msg_type returned: 2 for event 1|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Transport/0x0/context=(nil)|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Transport/0x0/gConnTab=0xf484290, addr=ISP-IP, port=5060, connid=2, transport=UDP|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0x0/Return existing connection for port 5060 connId 2|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0x0/Checking Invite Dialog|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0xb1b50c90/INVITE response with no RSEQ - disable IS_REL1XX|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_stop_timer: type=SIP_TIMER_TRYING value=500 retries=3|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/States/0xb1b50c90/0xb1b50c90 : State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_stop_timer: type=SIP_TIMER_EXPIRES value=180000 retries=0|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_start_timer: type=SIP_TIMER_EXPIRES value=180000 retries=0|2,100,230,1.4901096^ISP-IP^*
17:34:18.561 |EnvProcessUdpHandler::handle_input - handle = 334|*^*^*
17:34:18.561 |EnvProcessUdpHandler::handle_input Status: 0, Id: 2|*^*^*
17:34:18.561 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 396 from ISP-IP:[5060]:
[12623362,NET]
SIP/2.0 403 Forbidden
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
CSeq: 101 INVITE
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
Server: CISCO-SBC/2.x
Content-Length: 0
Contact: <sip:ISP-IP:5060>
[12623363,NET]
ACK sip:2484XXX@ISP-IP:5060 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Length: 0
INVITE sip:2484XXX@ISP's-Other-IP:5062 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Cisco-Guid: 0383266432-0000065536-0000191816-2219117834
Session-Expires: 1800
P-Asserted-Identity: <sip:057729XXXX@MY-CUCM-IP>
Remote-Party-ID: <sip:057729XXXX@MY-CUCM-IP>;party=calling;screen=yes;privacy=off
Contact: <sip:057729XXXX@MY-CUCM-IP:5062>
Max-Forwards: 68
Content-Type: application/sdp
Content-Length: 215
v=0
o=CiscoSystemsCCM-SIP 4052092 1 IN IP4 MY-CUCM-IP
s=SIP Call
c=IN IP4 MY-CUCM-IP
t=0 0
m=audio 29792 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
|2,100,56,1.173711431^MY-CUCM-IP^MTP_3
17:34:18.567 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 0|2,100,56,1.173711431^MY-CUCM-IP^MTP_3
17:34:18.567 |EnvProcessUdpHandler::fireSignal - SEND: index = 0, handler = 0xa6b4d7c0|*^*^*
17:34:18.567 |EnvProcessUdpPort::fireSignal - SEND, destination = ISP's-Other-IP:5062|*^*^*
17:34:18.567 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 1177, ISP's-Other-IP:5062)|*^*^*
17:34:18.569 |EnvProcessUdpHandler::handle_input - handle = 335|*^*^*
17:34:18.569 |EnvProcessUdpHandler::handle_input Status: 0, Id: 0|*^*^*
17:34:18.569 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 394 from ISP's-Other-IP:[5062]:
[12623365,NET]
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900;rport=5062
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
CSeq: 101 INVITE
Server: kamailio (3.3.1 (x86_64/linux))
Content-Length: 0
17:34:18.587 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 375 from ISP's-Other-IP:[5062]:
[12623366,NET]
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900;rport=5062
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>;tag=dc6a4ae7
CSeq: 101 INVITE
Reason: Q.850;cause=0;text="unknown"
Content-Length: 0
|2,100,230,1.4901099^ISP's-Other-IP^*
[12623367,NET]
ACK sip:2484XXX@ISP's-Other-IP:5062 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>;tag=dc6a4ae7
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Length: 0SIP/2.0 403 Forbidden error
If your router is sending a SIP/2.0 403 Forbidden error to the SIP server you are registered to, there is a good chance your router is blocking the incoming call due to the toll-faud prevention feature that was added to IOS version 15.1(2)T.
How to Identify if TOLLFRAUD_APP is Blocking Your Call
If the TOLLFRAUD_APP is rejecting the call, it generates a Q.850 disconnect cause value of 21, which represents ‘Call Rejected’. The debug voip ccapi inout command can be run to identify the cause value.
Additionally, voice iec syslog can be enabled to further verify if the call failure is a result of the toll-fraud prevention. This configuration, which is often handy to troubleshoot the origin of failure from a gateway perspective, will print out that the call is being rejected due to toll call fraud. The CCAPI and Voice IEC output is demonstrated in this debug output:
%VOICE_IEC-3-GW: Application Framework Core: Internal Error (Toll fraud call rejected):
IEC=1.1.228.3.31.0 on callID 3 GUID=F146D6B0539C11DF800CA596C4C2D7EF
000183: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/ccCallSetContext:
Context=0x49EC9978
000184: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/cc_process_call_setup_ind:
>>>>CCAPI handed cid 3 with tag 1002 to app "_ManagedAppProcess_TOLLFRAUD_APP"
000185: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/ccCallDisconnect:
Cause Value=21, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
The Q.850 disconnect value that is returned for blocked calls can also be changed from the default of 21 with this command:
voice service voip
ip address trusted call-block cause
How to Return to Pre-15.1(2)T Behavior
Source IP Address Trust List
There are three ways to return to the previous behavior of voice gateways before this trusted address toll-fraud prevention feature was implemented. All of these configurations require that you are already running 15.1(2)T in order for you to make the configuration change.
Explicitly enable those source IP addresses from which you would like to add to the trusted list for legitimate VoIP calls. Up to 100 entries can be defined. This below configuration accepts calls from those host 203.0.113.100/32, as well as from the network 192.0.2.0/24. Call setups from all other hosts are rejected. This is the recommended method from a voice security perspective.
voice service voip
ip address trusted list
ipv4 203.0.113.100 255.255.255.255
ipv4 192.0.2.0 255.255.255.0
Configure the router to accept incoming call setups from all source IP addresses.
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
Disable the toll-fraud prevention application completely.
voice service voip
no ip address trusted authenticate
Two-Stage Dialing
If two-stage dialing is required, the following can be configured to return behavior to match previous releases.
For inbound ISDN calls:
voice service pots
no direct-inward-dial isdn
For inbound FXO calls:
voice-port
secondary dialtone -
Does the route list gets reset when adding route pattern in version 9 of CUCM?
Hi All,
I created a new route pattern in call manager version 9 and then associated it to an existing route list. But when I was trying to save the route pattern. I was prompted with the message saying " Any updates to this Route Pattern automatically resets the associated gateway or Route List"
I was not able to see this message on CUCM version 8.6 however on CUCM version 9 it is there.
Does that mean that the route list/gateway will be reset and the existing calls will be dropped?
Or is it a bug?
A response will be greatly appreciated.
ThanksHi Farhad,
It does pop up when we modify the Route Pattern in CUCM 8.6 too.
If you asociate the GW/Trunk directly to Route Pattern, yes, it will reset them and calls will be interrupted. To avoid that, we need to associate the Route List to Route Patterns.
//Suresh
Please rate all the useful posts. -
Route Pattern CSV File Removes "0" from Called Party Prefix Digits Field
I want to upload over 350 route patterns using BAT Tool in CUCM 9.1. All patterns must have their Called Party Prefix Digits (Outgoing Calls) Field containing 10 numbers with the number "0" at the begining. The Problem is the CSV file removes the leading "0" form the digits. I tried to make the cell type in Excel as "Text" and it worked and the "0" is kept normally, but when I save and close the file then open it again, the cell is defaulted to "General" type and the "0" is disappeared again! Changing the CSV file format to any other one and uploading it to CUCM system generates an error stating that the file format is not supported.
Attached is a sample entry of the CSV file. I want to preserve the whole number "0541234567" in "PREFIX_DIGITS_CALLED_PARTY" Field.
Anyone can help me how to upload this big number of route patterns while preserving the number "0" at the begining?Make sure you change the csv file when using Excel to "Text" on the cell where the string starts with 0, otherwise Excel assumes this is a number and strips it.
HTH, please rate all useful posts!
Chris -
Route Pattern with Pause in CUCM 10.5
Dear Experts,
I need to create a route pattern with a pause to dial a secondary number.
the customer is forwarding international calls to tollfree number, then the user is dialling the international number.
He is requesting to make it as one step, dial the prefix and international number at the same time.
the pattern is 44.00!
44 will be transformed to the toll free number, and the rest is the international number.
Any idea
thank in advance
Regards
AnasJonathan is correct (+5) - Route pattern and neither the translation pattern would not be able to do that. Tried in my lab CUCM 10.5, it doesn't recognizes comma as a valid entry :
You can configure pause in the pots dial-peer by comma, but I doubt it will be helpful here.
-Terry -
Question About CVP Call Server Logs
Hi Members,
In the CallServer Logs i could see a [SIP_LEG_PRERTE_CORRID] - Publishing ,, [ICM_REQUEST_INSTRUCTION]
Could someone tellme what is the difference between [SIP_LEG] and [SIP_LEG_PRERTE_CORRID]
In what are all the scenarios i would get [SIP_LEG_PRERTE_CORRID] this leg
Regards,
SenthilHi Kris,
Thanks For your reply. Here is my Scenario.
PSTN---->VoiceGateway---->CUSP---->CVP CallServer----->ICM---->Script(Just to Play Music).
Here in ICM i have a Dialed Number configured and Mapped to a Calltype and Calltype is Associated in the Script where i have a IF Node (condition Match the Dialed Number).
We have lot of TFN's configured and everything is working fine without any issue.
Now i have to add new DID's in my Setup. When i test the Call it is not hitting my START NODE on the Script itself.
I have attached my ICM Router Logs here. Please let me know if you have any suggestion.
ICM Router Logs for the Dialog ID : 515149
07:21:29 rb-rtr Trace: Dialog (515149) has a correlation id (7 7) that is unknown.
07:21:29 rb-rtr Trace: For message (14) from routing client iyogi_CVP_PG_RC2 (ID 5003) could not find dialog id (7).
07:21:29 rb-rtr Trace: Router sending dialog fail reason (11) for dialog (515149).
CVP Call Server Logs for the Dialog ID : 515149
137620197: 10.110.10.18: May 27 2011 07:21:29.593 -0700: %_Message-6-com.dynamicsoft.DsLibs.DsUALibs.DsSipObject.Message: In DsMimeEntity.parseBody() - Content type application/x-q931 is unregistered.
137620198: 10.110.10.18: May 27 2011 07:21:29.593 -0700: %_Message-6-com.dynamicsoft.DsLibs.DsUALibs.DsSipObject.Message: In DsMimeEntity.parseBody() - Content type application/gtd is unregistered.
14036239: 10.110.10.18: May 27 2011 07:21:29.593 -0700: %CVP_8_0_ICM-7-CALL: {Thrd=pool-1-thread-50-ICM-6090032} CALLGUID = 31D4D479100001305602249B0A6E0A12, DLGID = -1 [null] - Processing ,, [MsgBus:NEW_CALL], ssId=SYS_SIP2, mediaType=, location=, locationpkid=, locationsiteid=, srcaddr=10.100.0.9, pstntrunkgroupid=10.100.0.9 , pstntrunkgroupchannelnum=2147483647, sipheader=, rckey=, rcday=, rcseq=, uui=, CallContext:, user.media.id: 31D4D479100001305602249B0A6E0A12,, LEGID = null, DNIS = -1, ANI = -1
14036240: 10.110.10.18: May 27 2011 07:21:29.593 -0700: %CVP_8_0_ICM-7-CALL: {Thrd=pool-1-thread-50-ICM-6090032} CALLGUID = 31D4D479100001305602249B0A6E0A12 - Correlation ID routed call
14036241: 10.110.10.18: May 27 2011 07:21:29.593 -0700: %CVP_8_0_ICM-7-CALL: {Thrd=pool-1-thread-50-ICM-6090032} CALLGUID = 31D4D479100001305602249B0A6E0A12, DLGID = 515149 [SIP_LEG_PRERTE_CORRID] - Publishing ,, [ICM_REQUEST_INSTRUCTION], dialogueId=515149, sendSeqNo=1, trunkGroupId=200, trunkNumber=1, serviceId=2, uui=, correlationId=7, location=, locationpkid=, pstntrunkgroupid=10.100.0.9 , pstntrunkgroupchannelnum=2147483647, sipheader=,, LEGID = 6B6C5D0A-87A311E0-8FF3C7B6-824C2EAB, DNIS = 35315260067, ANI = 18664281713
14036242: 10.110.10.18: May 27 2011 07:21:29.593 -0700: %CVP_8_0_ICM-7-CALL: {Thrd=pool-1-thread-138-ICM-6090033} CALLGUID = 31D4D479100001305602249B0A6E0A12, DLGID = 515149 [SIP_LEG_PRERTE_CORRID] - Processing ,, [ICM_DIALOGUE_FAILURE_EVENT], dialogueId=515149, sendSeqNo=1, errorCode = E_UNSPECIFIED_FAILURE,, LEGID = 6B6C5D0A-87A311E0-8FF3C7B6-824C2EAB, DNIS = 35315260067, ANI = 18664281713
14036243: 10.110.10.18: May 27 2011 07:21:29.593 -0700: %CVP_8_0_ICM-7-CALL: {Thrd=pool-1-thread-138-ICM-6090033} CALLGUID = 31D4D479100001305602249B0A6E0A12, DLGID = 515149 [SIP_LEG_PRERTE_CORRID] - Publishing ,, [MsgBus:DIALOGUE_FAILURE], ssId=SYS_SIP2, errorCode=E_UNSPECIFIED_FAILURE,, LEGID = 6B6C5D0A-87A311E0-8FF3C7B6-824C2EAB, DNIS = 35315260067, ANI = 18664281713
12185266: 10.110.10.18: May 27 2011 07:21:29.593 -0700: %CVP_8_0_SIP-3-SIP_CALL_ERROR: CALLGUID = 31D4D479100001305602249B0A6E0A12 LEGID = 6B6C5D0A-87A311E0-8FF3C7B6-824C2EAB - [INBOUND] - DIALOGUE_FAILURE from ICM Router sends 404 rejection to call. errorcode=15 [id:5004]
12185267: 10.110.10.18: May 27 2011 07:21:29.593 -0700: %CVP_8_0_SIP-3-SIP_CALL_ERROR: CALLGUID = 31D4D479100001305602249B0A6E0A12 LEGID = 6B6C5D0A-87A311E0-8FF3C7B6-824C2EAB - [INBOUND] - ABNORMALLY ENDING - SIP code [404], Reason Hdr [SIP;cause=404] Not Found, GW call using SURV TCL flag [false], NON NORMAL flag [true], USE ERROR REFER flag [true] with AGE (msecs) 0 and Call History : [id:5004]
Regards,
Senthil -
CVP call server logs
Hi All,
I am trying to figure out whether caller party(End User) hangup the call first or UCCE Agent.
Attaching CVP call server Logs& UCCE TCD& Route Call Details for your reference.From the CVP logs, it can be determined which side disconnected the call first. For each call, CVP keeps track each call leg. From Inbound Gateway to CVP is INBOUND leg, rest are OUTBOUND leg. You can then look at which leg the SIP BYE message is received first.
Since you have very basic log enabled, you will not see the exact SIP message. But it can be determined by the outcome of the message. Here is the snippet of the log during the disconnect:
Line 3766: 3083689: 10.180.245.43: Sep 12 2014 12:21:11.293 -0700: %CVP_8_5_SIP-7-CALL: {Thrd=DIALOG_CALLBACK.6} CALLGUID = CBCCDD8539E811E4A3E2CCEF48565980 LEGID = CC65CE04-39E811E4-87DFD7D1-64B198F2 - [INBOUND] DURATION (msecs) = 25610 - DIALOG TERMINATED. Reason: Q.850;cause=16
Line 3768: 3083690: 10.180.245.43: Sep 12 2014 12:21:11.293 -0700: %CVP_8_5_SIP-7-CALL: {Thrd=DIALOG_CALLBACK.6} Sending BUS MSG:>>HEADERS: (JMSType)=MsgBus:CALL_STATE_EVENT (JMSDestination)=Topic(CVP.SIP.CC.EVENT) (JMSTimestamp)=1410549671293 >>BODY: callguid=CBCCDD8539E811E4A3E2CCEF48565980 RouterCallKey=6472 RouterCallKeySent=true causecode=1 timezone=America/Los_Angeles RouterCallKeySequenceNumber=0 version=CVP_8_5 labeltype=1 RouterCallKeyDay=151099 calldate=Fri Sep 12 12:21:11 PDT 2014 label=190376 localOffset=-420 eventid=6 calllegid=CC65CE04-39E811E4-87DFD7D1-64B198F2 >>STATE: isTabular=false isWriteable=true cursor=-1
The first Termination message came on the INBOUND leg which is the PSTN. That means, PSTN side disconnected the call first.
Hope this helps.
Abu -
CUCM 7 Blackberry call forwarding
Hello,
I have CUCM 7. I wish to call forward to a blackberry device. When this is attempted the ISDN origin and destination cause codes are 4. This states a special information tone is to be generated. Does anyone know a solution for this ?
Many thanks
Stephen
Cause No. 4 - send special information tone [Q.850]
This cause indicates that the called party cannot be reached for reasons that are of a long
term nature and that the special information tone should be returned to the calling party.Hi, thanks for your response.
We need a SIP route pattern because the 3rd party server at the end of the SIP trunk uses SIP URI dialling when it talks back to the CallManager.
The requirements are that the call must first go over the SIP trunk to the 3rd party SIP server. If there is no response, then an alternate destination must be tried by the CallManager. This can be another SIP trunk to a second SIP server or it can be an DN internal to the CallManager.
Route groups can't be used because if a SIP trunk in configured and then assigned to a route group, the same SIP trunk can not then be referenced in a SIP route pattern.
In other words for communication to work with the 3rd party SIP server,
Step 1: The SIP trunk must be referenced directly by a route pattern.
Step 2: The SIP trunk must be referenced by a SIP route pattern.
We won't need route groups if the second destination is internal to the CallManager.
However, how do we hunt with 1 - Route pattern to SIP Trunk and then 2 - Internal DN? -
How do I add a route pattern to CUCM 7.1
I am currently using CUCM7.1 and need to add the route pattern 9911 to dial out to emergency dispatch. I do not want the capability of dialing 9 for an outside line for all users, just when I a calling 911 emergency. We recently changed to dialing a # for an outside line due to excessive 911 hangup calls. I tried adding 9911 to the route pattern list but I am missing something. I received the message stating could not complete call as dialed. Thank you, Cindy
Are you sure the phone has a CSS that allows you to dial 9911??
HTH
java
If this helps, please rate
www.cisco.com/go/pdihelpdesk -
Route pattern with called party transformations
HI All,
i wanto to add route pattern with transformation
i want to add RP with 9.001! predot
and want to convert to 9.01017! with called party transformations.
How we replace ! ? i've tried and it's error with message
Called Party Transform Mask - allowed characters are numeric (0-9),plus (+),asterisk (*),pound (#),X.
I've give screen shot for the configuration
Please anybody help.
Thanks in advance
Regards,
ATommyYou can create the RP with 9001.! and in the called party, discard predot and prefix 901017.
see below screenshot: -
CVP Call Studio Database Element
Hi all,
I got the below error when I tried to use the Database element in CVP Call Studio to connect to a MSSQL database.
touch111_24-7_Database,07/08/2014 13:52:31.724, The error was: A built-in element encountered an exception of type com.audium.server.AudiumException. There was a problem looking up the JNDI data source 'ivr1'. The root cause was: javax.naming.NameNotFoundException: Name ivr1 is not bound in this Context.
I have added the below to the context.xml file :
<Resource
name="jdbc/ivr1"
auth="Container"
type="javax.sql.DataSource"
driverClassName="com.microsoft.sqlserver.jdbc.SQLServerDriver"
url="jdbc:sqlserver://x.x.x.x:1433;databaseName=TestCallStudio;user=xxxx;password=xxxx"
/>
In the Database element settings in Call Studio, I put ivr1 in the JNDI Name field.
Can someone help me to know what could be causing this error to appear?
Thank you in advance.
Larachange jndi name in call studio from "ivr1" to "jdbc/ivr1"
and check if it is working or not.
regards
chintan -
Redirecting Voip call to another termination router
Hello All
The structure is very simple: There is one originating router (as5300 let say A) and
one terminating router (as5300 let say B) with 1 pri line. I want to route some calls whose have specific destination pattern from B to another router (let say C) to be terminated.
I have used "dial-peer voice xx voip" with "session target ipv4:x.x.x.x. on B.
But it failed. When I debug it on B ,it gives " can not gateway with C".
Do you think the idea is wrong for this structure . or how can I do this ( PS :I can administrate only termination routers )
thanks in advance..to tapatel:
First of all thank you for your assistance
Router B is not making call origination, it is only termination router.
Router A (I do not manage router A), is sending voip calls to B and I want to route some calls which has specific destination pattern from B to C. The configuration for B is like this:
! the dial-peer 101 and 99 below is working properly
dial-peer voice 101 pots
destination-pattern 3054T
port 0:1
prefix ,054
dial-peer voice 99 voip
destination pattern 3054T
session target ipv4:10.2.2.2 (the ip address of router A)
! I want to route it to router C with the lines below
dial-peer voice 102 voip
destination-pattern 30549238.....
session target ipv4:10.10.1.124 (this is the ip address of router C)
May be I have to use gatekeeper for this project but i am not sure if it is necessary.
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