Transfering Call in Cisco CME 10 7861 IP Phone
We have a issue in a 7861 IP Phone in CME version 10
We registered this IP Phone by configuring reference pool type 6961 under voice register global
CME version is 10 IOS is 15 3.3M4 in 2911
When we are dialing a number can we able to transfer the call before called party picks the call
This is because attended transfer is enabled by default. Turn it off using below commands and try again.
conf t
voice register template 1
no transfer-attended enable
Voice register pool X (your phone)
template 1
-Terry
Similar Messages
-
AutoAttendant not transferring calls in CUE 8.6 to CME 9.1
**Issue is calls are not transferring once extension or directory is pressed on the phone.
I upgraded our 2811 CME 7.1/CUE 7.0 router to 2911 15.2 CME 9.1/CUE8.6. It has CME and V/k9 license enabled. I uploaded the configuration used on the 2811 to the 2911. I added the ip addresses in the ip trusted list. Calls come in and out if I point the translation list to the phone and not to the auto attendant pilot number. Incase if it was toll fraud issue I disabled it but still a no go.
Everything worked fine on the 2811. I can't understand why it would not work on the 2911 15.2. Has anyone had any issues when upgrading from 12.4 to 15.x with auto attendant not transferring calls?
I ran a trace on the Cue when placed a call to extension 114:
20.10.10.1- CME interface address
20.10.10.5- CUE interface address
659 09/12 11:43:39.418 ACCN SIPS 0 Call.transferFailed(114, RESOURCE_NOT_ACKNOWLEDGING) SIPCallContact[id=33,type=Cisco SIP Call,implId=5738D3Dse-20-10-10-5# [email protected],active=true,state=CALL_ANSWERED,inbound=true,handled=false,locale=en_USOk I'm starting to think its a dtmf issue. I checked to see if I call from internal to the AA and dial an extension if it works but the same issue. I ran the debug voip ccapi inout and I see consume mask is not set. What would that indicate? Could that be the problem?
This is right when I dialed extension 114
2811-TEST#
001288: Sep 16 14:59:22.870: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_begin:
Consume mask is not set. Relaying Digit 1 to dstCallId 0x38
001289: Sep 16 14:59:22.870: //55/D62BA3D180C8/CCAPI/cc_relay_digit_begin_for_3way_conference:
Check DTMF relay digit begin for 3way conf
001290: Sep 16 14:59:22.870: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_end:
Consume mask is not set. Relaying Digit 1 to dstCallId 0x38
001291: Sep 16 14:59:22.870: //55/D62BA3D180C8/CCAPI/cc_relay_digit_end_for_3way_conference:
Check DTMF relay digit end for 3way conf
001292: Sep 16 14:59:23.194: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_begin:
Consume mask is not set. Relaying Digit 1 to dstCallId 0x38
001293: Sep 16 14:59:23.194: //55/D62BA3D180C8/CCAPI/cc_relay_digit_begin_for_3way_conference:
Check DTMF relay digit begin for 3way conf
001294: Sep 16 14:59:23.194: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_end:
Consume mask is not set. Relaying Digit 1 to dstCallId 0x38
001295: Sep 16 14:59:23.198: //55/D62BA3D180C8/CCAPI/cc_relay_digit_end_for_3way_conference:
Check DTMF relay digit end for 3way conf
001296: Sep 16 14:59:23.614: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_begin:
Consume mask is not set. Relaying Digit 4 to dstCallId 0x38
2811-TEST#
001297: Sep 16 14:59:23.614: //55/D62BA3D180C8/CCAPI/cc_relay_digit_begin_for_3way_conference:
Check DTMF relay digit begin for 3way conf
001298: Sep 16 14:59:23.618: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_end:
Consume mask is not set. Relaying Digit 4 to dstCallId 0x38
001299: Sep 16 14:59:23.618: //55/D62BA3D180C8/CCAPI/cc_relay_digit_end_for_3way_conference:
Check DTMF relay digit end for 3way conf
2811-TEST#
At this point theres silence on the phone I see this message:
2811-TEST#
001300: Sep 16 14:59:28.914: //55/D62BA3D180C8/CCAPI/ccGenerateToneInfo:
Stop Tone On Digit=FALSE, Tone=Null,
Tone Direction=Sum Network, Params=0x0, Call Id=55
001301: Sep 16 14:59:28.918: //56/D62BA3D180C8/CCAPI/cc_api_call_feature:
Feature Type=50, Interface=0x49FC2B80, Call Id=56
2811-TEST#
2811-TEST#
At this point I get the message "the phone number you are trying to reach" then the call disconnects
2811-TEST#
001242: Sep 16 14:48:57.719: %VOICE_IEC-3-GW: SIP: Internal Error (ACK wait timeout): IEC=1.1.129.7.67.0 on callID 52 GUID=5D21AC143CE711E480BCEEC6B624839
Also I checked show voice iec description:
2811-TEST#show voice iec description 1.1.129.7.67.0
IEC Version: 1
Entity: 1 (Gateway)
Category: 129 (Call setup timeout)
Subsystem: 7 (SIP)
Error: 67 (ACK wait timeout)
Diagnostic Code: 0
2811-TEST# -
Transfer VOIP Calls Between Cisco Desk Phone and Cisco Jabber For IPhone 9.5
Does anyone know how to transfer an active voip call from a Cisco IP Desk Phone to Cisco Jabber for IPhone? I can transfer a call from Cisco Jabber for IPhone to my Cisco IP Desk Phone no problem. I put the call on hold and then click "Resume" on my Cisco IP Desk Phone. However I cannot do the same but the other way around. If I put the call on hold on my Cisco IP Desk Phone, I see "no active call" on my Jabber client. The only information I could find slighlty relevant was using the Mobility Key/Remote Destination Profile feature however this defeats the object as this will forward to an external number, e.g. mobile and I just want to transfer the call within the VOIP environment between the two devices that are using the same directory number.
I am using Cisco Call Manager 9.1(2), Cisco Presence 9.1 and Cisco Jabber for IPhone 9.5.
Any help would be greatly appreciated.
Kind Regards,
Paul Parker.Did you ever find an answer to this ?
I am seeing the same behavior and trying so see if I can put calls on hold and pick them up both ways also.
The only answer I seem to have found is to use park instead
That would/should work but I would just prefer to hold/unhold
Just not sure why we would not be able to hold/unhold on what is essentially a "shared" line
Does anyone have this working for them ? -
Cisco CME and Calls through SIP provider
Hello, friends.
There are Cisco (C2801-ADVENTERPRISEK9_IVS-M), Version 15.1 (4) M7.
Telephones connected to SCCP, registered SIP from the provider.
When I try to call to test number 4444 through sip in debug I see:
*Feb 10 01:51:25.317: //53363/2739DFE79696/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP XXXXXXXXXXX:5060;branch=z9hG4bK100D02077;rport=5060
From: "TEST" <sip:[email protected]>;tag=131CC60C-1D40
To: <sip:[email protected]>;tag=b638310eda6e4a73cf10b7fe3c94c572.bef7
Call-ID: [email protected]
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="sip.zadarma.com", nonce="Uvf1OFL39Awnou/oMiaFQrf9jyybhFmf", qop="auth"
Server: kamailio (4.0.3 (x86_64/linux))
Content-Length: 0
*Feb 10 01:51:25.325: //53363/2739DFE79696/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP XXXXXXXXXX:5060;branch=z9hG4bK100D02077
From: "TEST" <sip:[email protected]>;tag=131CC60C-1D40
To: <sip:[email protected]>;tag=b638310eda6e4a73cf10b7fe3c94c572.bef7
Date: Sun, 09 Feb 2014 21:51:25 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
Cisco при этом зарегана у провайдера SIP
DC#show sip-ua register status
Line peer expires(sec) registered P-Associ-URI
Configuration:
voice service voip
ip address trusted list
ipv4 178.16.26.122 255.255.255.255
ipv4 144.76.42.108 255.255.255.255
ipv4 176.9.145.115 255.255.255.255
ipv4 5.9.108.25 255.255.255.255
ipv4 78.46.95.118 255.255.255.255
ipv4 89.249.23.194 255.255.255.255
ipv4 178.16.26.124 255.255.255.255
ipv4 176.9.85.133 255.255.255.255
ipv4 46.4.53.86 255.255.255.255
ipv4 5.9.84.165 255.255.255.255
ipv4 78.16.26.122 255.255.255.255
ipv4 77.235.62.222 255.255.255.255
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
registrar server
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
codec preference 3 g711alaw
voice register global
max-dn 10
max-pool 10
voice register dn 1
number 150
voice register dn 2
number 151
voice translation-rule 9
rule 1 /^95/ //
voice translation-rule 1020
rule 1 /^.$/ /40232/
voice translation-profile outgoing
translate calling 1020
translate called 9
mgcp fax t38 ecm
mgcp profile default
dial-peer voice 2 voip
translation-profile outgoing outgoing
destination-pattern 95....
session protocol sipv2
session target sip-server
voice-class codec 1
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay rtp-nte
no vad
sip-ua
credentials username 40232 password 7 XXXXXXXXXX realm sip.zadarma.com
authentication username 40232 password 7 XXXXXXXXXXXX realm sip.zadarma.com
registrar dns:sip.zadarma.com:5060 expires 3600
sip-server dns:sip.zadarma.com:5060
connection-reuse
host-registrar
DC#show sip-ua register status
Line peer expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
150 40001 12 no
40232 -1 550 yes
SIP provider says cisco trying to call with the internal call number, and it is necessary in order that have an SIP provider:
Wrong Remote-Party-ID: "Vankuver" <sip:61@<my ip>>;party=calling;
Should be so sip:40232@<my ip>
Please help me!Yes, I behind nat.
*Feb 10 18:11:53.425: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/TCP 192.168.11.14:42294;branch=z9hG4bK-d8754z-e645887cf7416a27-1---d8754z-;rport
Max-Forwards: 70
Contact:
To: "954444"
From: "150";tag=7b409f06
Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 314
v=0
o=- 2 2 IN IP4 192.168.11.14
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.11.14
t=0 0
m=audio 5724 RTP/AVP 107 0 8 101
a=alt:1 2 : gNONJ/Dj BaLJhmb/ 10.200.16.55 5724
a=alt:2 1 : DQ3e8qud c1qVrWui 192.168.11.14 5724
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
*Feb 10 18:11:53.477: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK1038E7FF
From: "" >;tag=169E6BC4-1E16
To: [email protected]>
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0541864002-2442400227-2618163141-2285537806
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1392041513
Contact: outside ip cisco cme:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 262
v=0
o=CiscoSystemsSIP-GW-UserAgent 8076 2450 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18534 RTP/AVP 0 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
*Feb 10 18:11:53.481: //54340/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.11.14:42294;branch=z9hG4bK-d8754z-e645887cf7416a27-1---d8754z-;rport
From: "150";tag=7b409f06
To: "954444"
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
*Feb 10 18:11:53.625: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP outside ip cisco cme:5060;branch=z9hG4bK1038E7FF;rport=5060
From: "" ;tag=169E6BC4-1E16
To: [email protected]>;tag=9fedfddccf3bcc4a1975d2cdb2a664b8.7066
Call-ID: [email protected]
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="sip.zadarma.com", nonce="Uvja/1L42dNbKQpCc2GzgagslkjyE1Pn", qop="auth"
Server: kamailio (4.0.3 (x86_64/linux))
Content-Length: 0
*Feb 10 18:11:53.633: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDPoutside ip cisco cme:5060;branch=z9hG4bK1038E7FF
From: "150" [email protected]>;tag=169E6BC4-1E16
To: [email protected]>;tag=9fedfddccf3bcc4a1975d2cdb2a664b8.7066
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
*Feb 10 18:11:53.637: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDPoutside ip cisco cme:5060;branch=z9hG4bK1038F25FC
From: "" ;tag=169E6BC4-1E16
To: [email protected]>
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0541864002-2442400227-2618163141-2285537806
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Timestamp: 1392041513
Contact: :5060>
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="40232",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="df38cd7f4af8e4a808fbbfdf5a7dd6a1",nonce="Uvja/1L42dNbKQpCc2GzgagslkjyE1Pn",cnonce="E701683F",qop=auth,algorithm=md5,nc=00000001
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 262
v=0
o=CiscoSystemsSIP-GW-UserAgent 8076 2450 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18534 RTP/AVP 0 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
*Feb 10 18:11:53.981: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK1038F25FC;rport=5060
From: "" ;tag=169E6BC4-1E16
To: [email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
Server: kamailio (4.0.3 (x86_64/linux))
Content-Length: 0
*Feb 10 18:11:54.385: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 92.63.X:5060;rport=5060;branch=z9hG4bK1038F25FC
Record-Route:
From: "k40232" ;tag=169E6BC4-1E16
To: [email protected]>;tag=as7e8de8e5
Call-ID: [email protected]
CSeq: 102 INVITE
Server: Zadarma Voip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 281
v=0
o=root 1942395501 1942395501 IN IP4 178.16.26.124
s=Asterisk PBX
c=IN IP4 178.16.26.124
t=0 0
m=audio 12164 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
*Feb 10 18:11:54.409: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.xxxx.xxxx:5060;branch=z9hG4bK10390E63
From: "150" [email protected]>;tag=169E6BC4-1E16
To: [email protected]>;tag=as7e8de8e5
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: [email protected]
Route:
Max-Forwards: 70
CSeq: 102 ACK
Proxy-Authorization: Digest username="40232",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="df38cd7f4af8e4a808fbbfdf5a7dd6a1",nonce="Uvja/1L42dNbKQpCc2GzgagslkjyE1Pn",cnonce="E701683F",qop=auth,algorithm=md5,nc=00000001
Allow-Events: telephone-event
Content-Length: 0
*Feb 10 18:11:54.429: //54340/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.11.14:42294;branch=z9hG4bK-d8754z-e645887cf7416a27-1---d8754z-;rport
From: "150";tag=7b409f06
To: "954444";tag=169E6F78-88E
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.
CSeq: 1 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: :5060;transport=tcp>
Supported: replaces
Server: Cisco-SIPGateway/IOS-12.x
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 193
v=0
o=CiscoSystemsSIP-GW-UserAgent 149 3396 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 17190 RTP/AVP 8
c=IN IP4 92.63.108.115
a=rtpmap:8 PCMA/8000
a=ptime:20
*Feb 10 18:11:54.653: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 91.231.141.230:42294;branch=z9hG4bK-d8754z-95374017c126c928-1---d8754z-;rport
Max-Forwards: 70
Contact:
To: "954444";tag=169E6F78-88E
From: "150";tag=7b409f06
Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.
CSeq: 1 ACK
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0 -
Cisco CME: calls through SIP-provider again
Hello,friends!
I have already published a discussion here https://supportforums.cisco.com/discussion/12089656/cisco-cme-and-calls-through-sip-provider and you helped me, everything works well for Russian numbers.
When I tried to add the configuration for calls to Belarus, again, there was a problem. I do not understand why, although the configuration ideintichnaya.
My config:
voice service voip
ip address trusted list
ipv4 178.16.26.122 255.255.255.255
ipv4 144.76.42.108 255.255.255.255
ipv4 176.9.145.115 255.255.255.255
ipv4 5.9.108.25 255.255.255.255
ipv4 78.46.95.118 255.255.255.255
ipv4 89.249.23.194 255.255.255.255
ipv4 178.16.26.124 255.255.255.255
ipv4 176.9.85.133 255.255.255.255
ipv4 46.4.53.86 255.255.255.255
ipv4 5.9.84.165 255.255.255.255
ipv4 78.16.26.122 255.255.255.255
ipv4 77.235.62.222 255.255.255.255
ipv4 81.88.86.11 255.255.255.255
ipv4 192.168.1.50 255.255.255.255
ipv4 217.150.198.44 255.255.255.255
ipv4 178.63.96.3 255.255.255.255
ipv4 178.63.96.28 255.255.255.255
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
sip
registrar server
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
codec preference 3 g711alaw
voice class sip-profiles 20
request INVITE sip-header From modify "\"(.*)\" <sip:(.*)@(.*)>" "\"\" <sip:[email protected]>"
voice translation-rule 9
rule 1 /^98/ /7/
voice translation-rule 10
rule 1 /^9/ //
voice translation-rule 1020
rule 1 /^.*$/ /141756/
voice translation-rule 1030
rule 1 /^.*/ /141756/
voice translation-rule 1040
rule 1 /^.*$/ /21/
voice translation-profile incoming
translate called 1040
voice translation-profile outgoing
translate calling 1030
translate called 9
voice translation-profile outgoing-mezhdunarod
translate calling 1030
translate called 10
voice-card 0
dial-peer voice 2 voip
description TO-RUSSIA
translation-profile outgoing outgoing
preference 1
destination-pattern 98..........
session protocol sipv2
session target sip-server
voice-class codec 1
no voice-class sip outbound-proxy
voice-class sip profiles 20
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay rtp-nte sip-notify
no vad
dial-peer voice 3 voip
translation-profile incoming incoming
incoming called-number 141756
voice-class codec 1
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay rtp-nte
no vad
dial-peer voice 4 voip
description To-Belarus
translation-profile outgoing outgoing-mezhdunarod
destination-pattern 9375.........
session protocol sipv2
session target sip-server
voice-class codec 1
no voice-class sip outbound-proxy
voice-class sip profiles 20
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay rtp-nte sip-notify
no vad
sip-ua
credentials username 141756 password 7<pass> realm sip.zadarma.com
authentication username 141756 password 7 <pass>
no remote-party-id
registrar 1 dns:sip.zadarma.com expires 3600
sip-server dns:sip.zadarma.com
connection-reuse
host-registrar
DEBUG ccsip message:
Jun 17 14:23:09.033: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK36571F65
From: "" <sip:[email protected]>;tag=40FCB218-23D7
To: <sip:[email protected]>
Date: Tue, 17 Jun 2014 09:23:09 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3536040247-4114026979-2924673736-0741251102
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1402996989
Contact: <sip:[email protected]:5060>
Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 309
v=0
o=CiscoSystemsSIP-GW-UserAgent 6656 8059 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18252 RTP/AVP 0 18 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Jun 17 14:23:09.089: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK36571F65;rport=5060
From: "" <sip:[email protected]>;tag=40FCB218-23D7
To: <sip:[email protected]>;tag=b638310eda6e4a73cf10b7fe3c94c572.6d40
Call-ID: [email protected]
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="sip.zadarma.com", nonce="U6AYAFOgFtT86kmu2Fr5tYxLYGEexIl1", qop="auth"
Server: kamailio (4.1.2 (x86_64/linux))
Content-Length: 0
Jun 17 14:23:09.169: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK36571F65
From: "Vankuver" <sip:[email protected]>;tag=40FCB218-23D7
To: <sip:[email protected]>;tag=b638310eda6e4a73cf10b7fe3c94c572.6d40
Date: Tue, 17 Jun 2014 09:23:09 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
Jun 17 14:23:09.169: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK365820F2
From: "" <sip:[email protected]>;tag=40FCB218-23D7
To: <sip:[email protected]>
Date: Tue, 17 Jun 2014 09:23:09 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3536040247-4114026979-2924673736-0741251102
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1402996989
Contact: <sip:[email protected]:5060>
Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="141756",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="9534322838cbf2e265b2004bc0aa240e",nonce="U6AYAFOgFtT86kmu2Fr5tYxLYGEexIl1",cnonce="FFF9A231",qop=auth,algorithm=md5,nc=00000001
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 309
v=0
o=CiscoSystemsSIP-GW-UserAgent 6656 8059 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18252 RTP/AVP 0 18 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Jun 17 14:23:09.637: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK365820F2
From: "" <sip:[email protected]>;tag=40FCB218-23D7
To: <sip:[email protected]>
Date: Tue, 17 Jun 2014 09:23:09 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3536040247-4114026979-2924673736-0741251102
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1402996989
Contact: <sip:[email protected]:5060>
Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="141756",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="9534322838cbf2e265b2004bc0aa240e",nonce="U6AYAFOgFtT86kmu2Fr5tYxLYGEexIl1",cnonce="FFF9A231",qop=auth,algorithm=md5,nc=00000001
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 309
v=0
o=CiscoSystemsSIP-GW-UserAgent 6656 8059 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18252 RTP/AVP 0 18 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Jun 17 14:23:10.621: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK365820F2
From: "" <sip:[email protected]>;tag=40FCB218-23D7
To: <sip:[email protected]>
Date: Tue, 17 Jun 2014 09:23:10 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3536040247-4114026979-2924673736-0741251102
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1402996990
Contact: <sip:[email protected]:5060>
Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="141756",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="9534322838cbf2e265b2004bc0aa240e",nonce="U6AYAFOgFtT86kmu2Fr5tYxLYGEexIl1",cnonce="FFF9A
All possible debugging has been turned off
DC#231",qop=auth,algorithm=md5,nc=00000001
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 309
v=0
o=CiscoSystemsSIP-GW-UserAgent 6656 8059 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18252 RTP/AVP 0 18 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Debug voice ccapi inout:
Destination Pattern=9375........., Called Number=375298911396, Digit Strip=FALSE
Jun 17 15:22:13.073: //14425/13366763AF35/CCAPI/ccCallSetupRequest:
Calling Number=141756(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=375298911396(TON=Unknown, NPI=Unknown),
Redirect Number=, Display Info=Vankuver
Account Number=, Final Destination Flag=FALSE,
Guid=13366763-F540-11E3-AF35-FAC82C2E981E, Outgoing Dial-peer=4
Jun 17 15:22:13.073: //14425/13366763AF35/CCAPI/cc_api_display_ie_subfields:
ccCallSetupRequest:
cisco-username=
----- ccCallInfo IE subfields -----
cisco-ani=141756
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=0
dest=375298911396
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=0
cisco-rdnsi=0
cisco-redirectreason=0 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
Jun 17 15:22:13.073: //14425/13366763AF35/CCAPI/ccIFCallSetupRequestPrivate:
Interface=0x6968AA04, Interface Type=3, Destination=, Mode=0x0,
Call Params(Calling Number=141756,(Calling Name=Vankuver)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=375298911396(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=RegularLine, FinalDestinationFlag=FALSE, Outgoing Dial-peer=4, Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
Jun 17 15:22:13.073: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jun 17 15:22:13.073: :cc_get_feature_vsa malloc success
Jun 17 15:22:13.073: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jun 17 15:22:13.077: cc_get_feature_vsa count is 2
Jun 17 15:22:13.077: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jun 17 15:22:13.077: :FEATURE_VSA attributes are: feature_name:0,feature_time:1819298856,feature_id:3371
Jun 17 15:22:13.077: //14427/13366763AF35/CCAPI/ccIFCallSetupRequestPrivate:
SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
Jun 17 15:22:13.077: //14427/13366763AF35/CCAPI/ccCallSetContext:
Context=0x6C726BF4
Jun 17 15:22:13.077: //14425/13366763AF35/CCAPI/ccSaveDialpeerTag:
Outgoing Dial-peer=4
Jun 17 15:22:13.085: //14427/13366763AF35/CCAPI/cc_api_call_proceeding:
Please help me... I don't know what to do!You need to contact service provider for this , after authentication challenge your sip provider is not sending any response.
Contact them and ask whether they had received INVITE with proxy authentication details or not. -
Login to cisco CME as Administrator failed check your call manager express
Hi Experts
CME and CUE in one router. when i access the CUE from the IE , i put the CUE username and password and i get in. After that it asks me to enter CME username and password to run the wizard. whenever i put the password i get this crappy message "Login to cisco CME as Administrator failed. check your call manager express config" i have cheked my config many times. Please let me know if someone has faced this problem or any suggestion on this. today is the 2nd time i have faced this problem , last time I cud not solve it and end up wasting 6 hours...
please helpHi friend,
Here is the Prerequisites for Installing Cisco Unity Express Software. As David describes, may be the CME admin account is missing:
http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/rel3_1/installation/guide/prereq31.html#wpmkr1112912
Try this, and let us know.
Best regards,
- Adrián. -
I have a client who liked Cisco initially but now likes Nortel IP-PBX.
I dont know much about Nortel to compare it to Cisco's CME.
Can anyone please provide me some links or information on why Cisco CME solution is better than Nortel's Hybrid solution.
Thanks
ADWhat about applications? Is Nortel BCM Proprietary? Does it allow thir party integration for Phone applications?
Our main concerns are:
1. Management of the system.
2. Reliability.
3. Need for third party application integration.
4. What is the value of BCM vs CME?
Our company has 120 phones at the moment. But that number will increase to 180 in next 2 years. We may not reach 240 in next 5-7 years. So we are condidering call Manager express solution. Initially we were looking at Call Manager solution but got some bad feedback about the CCM. Is CCM 5.0 reliable?
What does CCM provide that CME does'nt ?
We will be adding 15 warehouse locations in the future. But each location will have 2 users. There will be 2 Analog lines for these locations and may be use Softphone through VPN tunnel. We do not have point to point T-1's for remote locations. Just a Internet T-1 for VPN tunnel to main location.
Thanks for your help
AD -
Cisco VG 224 Analog Phone Gateway
Hi
Can i use cisco vg224 gateway in a scenario like -
cisco ISR+CCM express -->WAN LINK --> cisco ISR -> VG224 gateway --> analog phones.
If this scenario is good then, can i do call transfer and forwarding with analog phones with any extensions digital/analog on that setup?
thanks
regards
Rakesh
=====The Cisco VG 224 is an analog voice gateway that manages up to 24 FXS ports for analog devices. Call control of the analog FXS ports is provided by Cisco CallManager or by a Cisco CallManager Express (Cisco CME) system. An SCCP telephony control (STC) application on the Cisco VG 224 functions as a a proxy to translate call-control messages between the call-control system (Cisco CallManager or Cisco CME) and the voice gateway.
This URL should help you:
http://www.cisco.com/en/US/products/ps6441/products_feature_guide09186a0080483a76.html
http://www.cisco.com/en/US/products/ps6441/products_feature_guide09186a0080483a76.html -
Hi to all,
I have a router Cisco 2801 (is a lab scenary) configured as call control. I have configured a CME 8.0. All is working properly but when I put a call waiting with an analogue phone connected to an ATA, MoH works fine but when I put the call waiting with a 7911 IP phone, the MoH isn't ear in the other peer.
Any Idea?? Seems something about DSP but, why works when I hold the call with the ATA??
The two types of phones are configured by the same mode. Tell me if you need something about.
Thanks for your help.Hi,
May you please attach the out put of show voice dsp group all, show voice dsp detail for a working and non working call along with the output of debug ephone moh and debug voice ccapi inout for both the calls along with the calling, called and the party which pressed the hold key. Also send me the show tech from the router
Regards
Himank -
Cisco CME CfwdAll and Night Service
Hi,
we would like to forward a normal DN to a huntgroup (200 in the below example), but when the night-service is active,
the calls should be forwarded to a different number (300 in the below example):
ephone-dn 10 dual-line
number 100
call-forward all 200
call-forward night-service 300
However, it seems that "call-forward all" has priority over "call-forward night-service", so it won´t work like intended.
Any ideas how this could be achieved?
Many thanks
HeinzHave you verified your night service is configured correctly and it works if call fwd all is not set? The below document says call-fwd night service should have precedence over call-fwd all:
Ref CME Admin guide: http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeadm/cmetrans.html#pgfId-1106036
Call Forwarding
Call forwarding diverts calls to a specified number under one or more of the following conditions:
All calls—When all-call call forwarding is activated by a phone user, all incoming calls are diverted. The target destination for diverted calls can be specified in the router configuration or by the phone user with a soft key or feature access code. The most recently entered destination is recognized by Cisco Unified CME, regardless of how it was entered.
No answer—Incoming calls are diverted when the extension does not answer before the timeout expires. The target destination for diverted calls is specified in the router configuration.
Busy—Incoming calls are diverted when the extension is busy and call waiting is not active. The target destination for diverted calls is specified in the router configuration.
Night service—All incoming calls are automatically diverted during night-service hours. The target destination for diverted calls is specified in the router configuration.
A directory number can have all four types of call forwarding defined at the same time with a different forwarding destination defined for each type of call forwarding. If more than one type of call forwarding is active at one time, the order for evaluating the different types is as follows:
1. Call forward night-service
2. Call forward all
3. Call forward busy and call forward no-answer
-Terry
Please rate all helpful posts -
Unity Connection transferring calls to CUCM
I have posted here before, continuing my learning in my lab.
CUCM 9 and CUC 9. 4 IP Phones registered and all have mailboxes. I have a System Call Handler configured to answer calls coming from a CTI Route Point.
I have been playing with a scenario trying to learn all the capabilities of the various call handlers.
Im currently trying to allow the caller to transfer to a phone number that is registered in CUCM - but does NOT have a mailbox.
My CUC Configuration
Call Handler Basics
Callers Hear -> My Personal Recording
During Greeting -> Allow Transfers to Numbers Not Associate with Users or Call Handlers
After Greeting -> Conversation -> Caller System Transfer
System Settings -> Restriction Tables
Default Transfer
I added a new entry and specifically entered the DN that I was trying to transfer to (123)
I unchecked the "blocked" for all entries
Call Management -> Call Routing -> Forwarded Routing Rules
Attempt Forward Routing Rule -> Send Call to -> Conversation -> Attempt Forward
From an IP Phone, I dial the CTI RP and get forwarded to the main / opening call handler. I traverse a couple of menu tree levels to get to my call handler where I am attempting to transfer back to CUCM. I hear my recorded personal greeting instructing me to enter my number. Whether I wait for the recording to end or whether I enter the digits during the greeting, the results are the same. The result is that nothing happens. No recorded message telling me that I cant be transferred to that number - nothing at all. The call just terminates.
I started up Connection Remote Port Status monitor and I can watch the sequence of the call. When I enter the digits of the target phone (123), I see in the Remote Port Status Monitor that it sees and receives the DTMF digits, but then nothing happens.
I next tested by adding a test user in CUC with the same DN of 123. As soon as I did that, I repeated the attempted call and when I dialed "123", I get the recorded message "Wait while I transfer that call" and then the call completes successfully.
This is my home lab so I can make any changes that you suggest with no user implications.
JeffHi Jeff,
That is weird! It is still shown in this 9.x doc
Overview of Default Restriction Tables in Cisco Unity
Connection 9.x
Default Fax
Restricts numbers for fax delivery.
Default Outdial
Restricts numbers for message notifications
Default System Transfer
Restricts numbers that can be used for
Caller system transfers, which allow
unidentified callers to transfer to a
number that they specify. For example,
callers may want to dial a lobby or conference room phone that is not
associated with a Connection user. By default, the table does not allow
Connection to dial any numbers.
Default Transfer
Restricts numbers for call transfers.
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/9x/administration/guide/9xcucsagx/9xcucsag110.pdf
Cheers!
Rob
"May your heart always be joyful
And may your song always be sung
May you stay forever young "
- Dylan -
We are using CUCM 8.6.2:
User-A, with extension ‘xxxxx’ calling internal switchboard operator ‘yyyyy’ (switchboard operator using ARC operator console system) with the request to connect to doctor’s mobile number ‘zzzzzzzzzz’;
Once the switchboard operator transferred (blind or consulted all the same) the call to doctor’s mobile ‘zzzzzzzzzz’, User-A can see the doctor's mobile 'zzzzzzzz' on his/her phone ‘xxxxx’.
How can I restrict User-A’s phone with extension ‘xxxxx’ seeing doctor’s mobile ‘zzzzzzzzzz’ after being transferred by switchboard operator ‘yyyyy’.
I would be very greatful to get some hlep on this matter.
Early thanks to all of you who has time, clue or solution for this.
Regards,
MohibI ended up logging a call with Cisco and got the solution, so thought I'll share with you all.
Although there is possible configuration per DN but that can be a bit of complex configuration and also will vary depend upon individual route pattern and circumstances.
I used the global configuration suggested by Cisco engineer for our organisation as mentioned bellow:
Choose Service > Service Parameters in the Cisco Unified CallManager Administration page.
Choose the PUB IP and then choose the Cisco CallManager service.
In Clusterwide Parameters (Device - General) change the parameter value of Always Display Original Dialed Numberto true. The default setting is false. -
SIP to SIP Call Failures on CME to CME - sip-ua conflict/issue?
Hi,
I have two existing CME systems which I wish to allow internal calls between. These calls will go over an IPSec VPN. However the calls are failing.
Phones DN22xx - London CME 2801 - PIX505 --- Internet ---ASA5505 - India CME 2801 - Phones DN400x
I have configured dial peers on both CME's and the IPSec VPN. I can ping between both systems. The VPN allows traffic between the interface IP's of the CME systems only.
London CME (local SCCP phones 22xx):
interface FastEthernet0/0.100
encapsulation dot1Q 100 native
ip address 10.0.10.250 255.255.255.0
voice class codec 101
codec preference 1 g729r8
codec preference 2 g711ulaw
codec preference 3 g711alaw
dial-peer voice 25 voip
description *** SIP Peer to India ***
answer-address 400.
destination-pattern 400.
voice-class codec 101
session protocol sipv2
session target ipv4:192.168.15.10
incoming called-number 400.
no vad
India CME (Local SSCP phones 400x):
interface FastEthernet0/0
ip address 192.168.15.10 255.255.255.0
voice class codec 100
codec preference 1 g729r8
codec preference 2 g711ulaw
codec preference 3 g711alaw
dial-peer voice 10 voip
description *** SIP Peer to London UK ***
answer-address 22..
destination-pattern 22..
voice-class codec 100
session protocol sipv2
session target ipv4:10.0.10.250
incoming called-number 22..
no vad
The CME system at India also has an existing SIP dial peer to a service provider and has sip-ua configured (username, password, realm and registrar).
A call from India (4005) to London (DN2207) fails, the ccsip debug attached. I'm assuming its because the sip-ua configuration is being used for these calls to when I don't want it to be. The from field shows âFrom: <sip:[email protected]â when I need this to be the internal IP 192.168.15.10.
Can anyone offer any assistance with this?
Regards,
ChrisHi,
thanks for your input however thats not the problem. 201.196.128.56 isn't an address on the router, it only has one IP and its 192.168.15.10.
The 201.196.128.56 address is the NAT'd address on the firewall. So that when a SIP call is made to the internet with sip-ua the from address is the public IP.
Chris -
CTI OS desktop client freezes after release of transferred call
Using CTI OS desktop v7.5(9), UCCE 7.5(9), Endpoint is CIPC 8.6.3 on CUCM 7.1.3
ACD call received by Agent 1 on ext 1111
Agent 1 transfers to 1112
Agent 2 answers transferred call on 1112 using answer call button.
Agent 2 disconnects using Release Call button on CTI OS desktop
Call disconnects but stays in grid with status 'Talking' and buttons on CTI OS desktop become unresponsive. CIPC is fully usable independently.
Agent 2 can reset by killing the CTIOS Desktop process and restarting.
This condition ONLY occurs as the result of an inbound transfer to Agent 2. It does not appear to happen when a call is transferred to Agent 1 in a similar scenario (Agent 1 is on a 7942)
Any thoughts as to why this occurs?
Thanks so much for any assistance!Hi,
looks like for some reason the CIPC does not informs the CUCM about the call released event. Or it does, it's only the JTAPI gateway that misses it.
Did you try upgrading/downgrading the firmware of the CUPC?
Also, I am not quite sure that CIPC 8.x is supported. It might be, but it certainly is not on the list:
http://docwiki.cisco.com/wiki/Unified_CCE_Software_Compatibility_Matrix_for_ALL_7.x_Releases
G. -
Cisco CME integration with NICE recorder
Hi Team,
Please let me know if we can integrate Cisco CME with NICE recorder. If yes, please share the steps involved in configuration on CME side and NICE.
Warm Regards,
Dinesh RathiHi Anas,
If the panasonic prtocol is H323 will be no problem the CME dial peer will configure as below:-
dial-peer voice 121 voip
translation-profile outgoing prefix
destination-pattern x..
session target ipv4:192.x.x.x
dtmf-relay h245-alphanumeric h245-signal
but if the panasonic protocol is SIP the configuration on CME will be as below:-
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
registrar server expires max 250 min 200
no call service stop
dial-peer voice 2000 voip
destination-pattern y...
session protocol sipv2
session target ipv4:172.x.x.x
dtmf-relay sip-notify
codec g711ulaw
no vad
So please can you advise for the above if the two options are right or not??
Thanks
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