Configure Cisco CME auto-attendant to forward call after ext is dialed
I am trying to configure my CME if the auto attendant picks up and a users extension is dialed after hours to forward that call to the number configured for that extension, currently if the extension is dialed the call is not forwarded.
Thank you,
-Tom
Hi,
I have same or similar issue, when a person call my company and press option 3 for example (after hour) the call get forward to a manager mobile number, my question is how can I disable the forwarding from CUCM? the OS i use is
8.6.2.22900-9
Many thanks
Similar Messages
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Auto attendant not forwarding calls to Receiption
Hello,
I am having CME configured in Cisco 2821 router having CUE 3.2 and CME 7.0 installed. When I dial the call from outside the organization,I am able to hear the prompt but when I press 0 to to reach the reception I used to get prompt message as "Your call can not be completed" and then it plays another prompt.
Can anyone help here?Yes receiptionist phone is located at same site as CUE.
Customer has moved to new location and is a single site but now the PRI got changed, now they are using TTML.
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UC320 Auto Attendant answering delayed calls to fast.
I recently deployed a UC320W with the latest firmware upgrade (2.3.2). My trouble is the auto-attendant intermittently answers delayed ringing calls on the first or second ring no matter what the timer is set to (which is 25 sec.). Doesn't seem to be any particular line. There are 4 FXOs on the UC320 and a 5th FXO on a SPA8800. They are all configured as shared FXOs in key system fassion. I've done impedance matching on all 5 FXO lines.
This has happened twice before and I was able to workaround by downgrading firmware to 2.2.2.
I can't do that in this case, however, because a few of my phones are SPA512Gs, which, it's my understanding, only work with the 2.3.2 firmware.
Right now, I simply have the delayed ringing disabled.
We have another technician in another state that tells me he has run into the same issue as well.
Any suggestions?Hi Carlos,
Please clarify the issue you are having.
My understanding is that you have 5 FXO lines configured as shared FXO lines and call is forwarded if no answer to AA, timer is set to 25 seconds. AA answers the call intermittently on the first or second ringing before 25 seconds timer expries.
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Wendy -
hello
I have 2 Cisco 2801 running CME 4.3 and will be converter to SRST Routers, but they''ll have T1 card, because they used to has FXO card with 4 lines now they will have T1, but I wonder witch solution for auto attendant should I have for those routers especially the Unity Express.
ThanksBefore switching to CM, think twice, because you will gain a lot of complications, consts, and loose nice features and flexibility if you do that.
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Does anyone configure cisco router with MGCP to link Call agent Clarent ?
hi,
We require to configure As5300 with MGCP to link Clarent call agent. Does anyone have cisco router configuration ?
thanks.
best regards.
fred.Below is the sample configuration for the 5300 to Call-Agent. This is also dependant on which package is configured on the call-agent so we can configure it accordingly. Hope this helps.
version 12.3
no service pad
service timestamps debug datetime msec localtime show-timezone
service timestamps log datetime msec localtime show-timezone
service password-encryption
hostname AS5300-5
boot system tftp c5300-is-mz.123-2.T1 171.68.191.135
logging buffered 100000 debugging
enable password xxxx
backhaul-session-manager
set bh5300-vsc1 client nft
group bhgrp1 set bh5300-vsc1
session group bhgrp1 172.16.20.35 7007 172.16.20.28 7007 0
isdn switch-type primary-ni
isdn voice-call-failure 0
no scripting tcl init
no scripting tcl encdir
voice call carrier capacity active
voice class codec 1
codec preference 1 g723r63
codec preference 2 g711ulaw
no voice hpi capture buffer
no voice hpi capture destination
dial-control-mib retain-timer 240
dial-control-mib max-size 600
controller T1 0
framing esf
clock source line primary
linecode b8zs
pri-group timeslots 1-24 service mgcp
controller T1 1
framing esf
clock source line secondary 1
linecode b8zs
ds0-group 0 timeslots 1-24 type none service mgcp
controller T1 2
framing esf
clock source line secondary 2
linecode b8zs
controller T1 3
framing esf
clock source line secondary 3
linecode b8zs
interface Ethernet0
no ip address
no ip mroute-cache
shutdown
interface Serial0
no ip address
no ip mroute-cache
shutdown
clockrate 2015232
no fair-queue
interface Serial1
no ip address
no ip mroute-cache
shutdown
clockrate 2015232
no fair-queue
interface Serial2
no ip address
no ip mroute-cache
shutdown
clockrate 2015232
no fair-queue
interface Serial3
no ip address
no ip mroute-cache
shutdown
clockrate 2015232
no fair-queue
interface Serial0:23
no ip address
isdn switch-type primary-ni
isdn bind-l3 backhaul bh5300-vsc1
no cdp enable
interface FastEthernet0
ip address 172.16.20.28 255.255.255.192
no ip mroute-cache
duplex full
speed auto
no cdp enable
ip classless
ip route 0.0.0.0 0.0.0.0 172.16.20.1
no ip http server
radius-server host 172.21.59.165 auth-port 1645 acct-port 1646
radius-server key xxxxxxxx
radius-server vsa send accounting
voice-port 0:23
voice-port 1:0
mgcp
mgcp call-agent 172.16.20.35 2427 service-type mgcp version 0.1
mgcp quarantine mode loop
mgcp package-capability dtmf-package
mgcp package-capability rtp-package
mgcp package-capability as-package
mgcp default-package gm-package
mgcp profile default
timeout tsmax 100
no max1 lookup
dial-peer cor custom -
Auto attendant intermittently routes call to out of region/not in dial plan UM server
Hi all,
Exchange 2013 on prem, hardware not virtual. CU5 w/Lync 2013
I've got calls that get intermittently routed to UM servers that are out of region and not in the dial plan. The out of region UM server sees the call is outside of business hours & sends helpdesk calls to voicemail instead of the appropriate phone
menu.
Additionally, when Exchange admins who are in different time zones look at the GUI w/the AA's business hours they see a time skew even though the time displayed is listed as Eastern. I think the mis-routing & the time zone skew are related. When
the Tokyo server gets the call it checks the time: 3AM? Not in business hours even though in Eastern Time where the call is supposed to go, it is in business hours.
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“ms-diagnostics:
15032;reason="Re-directing request to the destination in 302”
Additionally the time zone on the AA schedule is set to Eastern Time. Why is the TYO UM server ignoring this and applying local time?
Any tips to point me in the right direction would be appreciated.
AdamNumber two was correct! The affected site did not have an arbitration mailbox. Details follow.
I still have the underlying problem of AA's getting the time zone of the UM server applied rather than the time zone they are allegedly set to (for example Beijing business hours served from a TYO UM server getting TYO time).
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It turns out that every AD site with Exchange servers needs to have an arbitration mailbox with the grammar generator role set & ready. If a site with UM servers does not have an arbitration mailbox it will proxy the call to another site that does.
In our case, it would route them to our Tokyo site that applied the wrong hours to the auto attendant.
Here's how we created the arbitration mailbox
[PS] C:\temp\autoattendant>New-Mailbox -Arbitration -Name "A new UM Grammar Mailbox" -Database <some db hosted in site> -UserPrincip
alName [email protected] -DisplayName "A new UM Grammar Mailbox"
C:\temp\autoattendant>Set-Mailbox [email protected] -Arbitration -UMGrammar:$true
This keeps the call from going out of site to an Exchange UM server in a different time zone.
The tricky bit is that this does not immediately work. The mailbox needs to pick up the OrganizationCapabilityUMGrammarReady capability which it will only get when the grammar generator runs. In 2010 you were able to kick this off manually. In
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Cisco TCL auto attendant without dial extension number
Hello, Im trying to use the TCL for auto attendant from the file its-Cisco.2.0.1.0.tcl but I need just the auto attendant prompt and not the extension number. At this moment I can do that but I listen two prompts (en_welcome.au and en_enter_destination.au) and I need just to listen the first one.
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RodrigoHello, Im trying to use the TCL for auto attendant from the file its-Cisco.2.0.1.0.tcl but I need just the auto attendant prompt and not the extension number. At this moment I can do that but I listen two prompts (en_welcome.au and en_enter_destination.au) and I need just to listen the first one.
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How to restart CME Auto Attendant ?
What command can reload AA but do not reload CME?
When change a CME on AA configure or change the AA en_welcome.au file.
application
service aa flash:its-CISCO.2.0.2.0.tcl
param operator 8000
paramspace english language en
paramspace english index 0
paramspace english location flash:
paramspace english prefix en
param aa-pilot 8000
param max-extension-length 4
param welcome-prompt en_welcome.auHere you are:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html#wp1055309
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Call forward to external number which has auto attendant
Hi
I am a voice administrator in my company
I want to forward all of my calls to my Other location's number.
Other location has only POTS line and to reach my extension user needs to go through Auto attendant menu.
Is it possible to enter a digit pattern in call forward destination in CUCM so that it can take care of Auto attendant menu of my Other location and land on my number?
We have CUCM 8 running.
Please help!!
AshwinHello Ashwin,
"Other location has only POTS line and to reach my extension user needs to go through Auto attendant menu."
What type of phone system and voice mail is providing the auto attendant?
How are the POTS lines(analog only correct?) terminated into the phone system? -
All off-silte call directly goes to Auto Attendant
Hello everyone,
I have an issue with UC520. There is one PSTN line connected to the voice port 0/2/0, All dial out works fine, All off-site calls goes directley to the Auto Attendant, however, interal dial-in works fine, I mean user can dial internal extension properly but not from offsite to insite.
I was wondering if any one can help me.
Here is the partal UC configuration:
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2014.01.13 13:51:51 =~=~=~=~=~=~=~=~=~=~=~=
sh run
Building configuration...
Current configuration : 31685 bytes
dot11 syslog
dot11 ssid uc520-data
vlan 1
authentication open
dot11 ssid uc520-voice
vlan 100
authentication open
ip source-route
ip cef
ip dhcp relay information trust-all
ip dhcp excluded-address 10.1.1.1 10.1.1.10
ip dhcp excluded-address 192.168.10.1 192.168.10.10
ip dhcp pool phone
network 10.1.1.0 255.255.255.0
default-router 10.1.1.1
option 150 ip 10.1.1.1
ip dhcp pool data
import all
network 192.168.10.0 255.255.255.0
default-router 192.168.10.1
ip name-server 63.203.35.55
ip inspect WAAS flush-timeout 10
ip inspect name SDM_LOW dns
ip inspect name SDM_LOW ftp
ip inspect name SDM_LOW h323
ip inspect name SDM_LOW https
ip inspect name SDM_LOW icmp
ip inspect name SDM_LOW imap
ip inspect name SDM_LOW pop3
ip inspect name SDM_LOW netshow
ip inspect name SDM_LOW rcmd
ip inspect name SDM_LOW realaudio
ip inspect name SDM_LOW rtsp
ip inspect name SDM_LOW esmtp
ip inspect name SDM_LOW sqlnet
ip inspect name SDM_LOW streamworks
ip inspect name SDM_LOW tftp
ip inspect name SDM_LOW tcp
ip inspect name SDM_LOW udp router-traffic
ip inspect name SDM_LOW vdolive
no ipv6 cef
multilink bundle-name authenticated
stcapp ccm-group 1
stcapp
stcapp feature access-code
stcapp supplementary-services
port 0/0/0
fallback-dn 301
port 0/0/1
fallback-dn 302
port 0/0/2
fallback-dn 303
port 0/0/3
fallback-dn 304
trunk group ALL_BRI
translation-profile outgoing PROFILE_ALL_BRI
voice call send-alert
voice rtp send-recv
voice service voip
sip
no update-callerid
voice class codec 1
codec preference 2 g729r8
voice class custom-cptone CCAjointone
dualtone conference
frequency 600 900
cadence 300 150 300 100 300 50
voice class custom-cptone CCAleavetone
dualtone conference
frequency 400 800
cadence 400 50 200 50 200 50
voice register global
max-dn 56
max-pool 14
voice translation-rule 4
rule 15 /^...$/ /0354434848/
voice translation-rule 1000
rule 1 /.*/ //
voice translation-rule 1111
voice translation-rule 1112
rule 1 /^0/ /*/
voice translation-rule 2222
voice translation-profile CALLER_ID_TRANSLATION_PROFILE
translate calling 1111
voice translation-profile CallBlocking
translate called 2222
voice translation-profile OUTGOING_TRANSLATION_PROFILE
translate called 1112
voice translation-profile PROFILE_ALL_BRI
translate calling 4
voice translation-profile nondialable
translate called 1000
voice-card 0
dspfarm
dsp services dspfarm
license udi pid UC520W-8U-2BRI-K9 sn FHK131827A2
archive
log config
logging enable
logging size 600
hidekeys
username cisco privilege 15 secret 5 $1$TC0B$LXMORw4u1vQpD/2eJdN4W1
username admin privilege 15 password 0 admin
username parham privilege 15 password 0 parham
ip tftp source-interface Loopback0
translation-rule 22
bridge irb
interface Loopback0
description $FW_INSIDE$
ip address 10.1.10.2 255.255.255.252
ip access-group 101 in
ip nat inside
ip virtual-reassembly in
interface FastEthernet0/0
description $FW_OUTSIDE$
ip address dhcp
ip access-group 104 in
ip nat outside
ip inspect SDM_LOW out
ip virtual-reassembly in
duplex auto
speed auto
interface Integrated-Service-Engine0/0
description cue is initialized with default IMAP group
ip unnumbered Loopback0
ip nat inside
ip virtual-reassembly in
service-module ip address 10.1.10.1 255.255.255.252
service-module ip default-gateway 10.1.10.2
interface FastEthernet0/1/0
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/1
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/2
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/3
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/4
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/5
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/6
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/7
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/8
switchport mode trunk
no ip address
macro description cisco-switch
interface BRI0/1/0
no ip address
isdn point-to-point-setup
isdn incoming-voice voice
isdn sending-complete
interface BRI0/1/1
no ip address
isdn point-to-point-setup
isdn incoming-voice voice
isdn sending-complete
interface Dot11Radio0/5/0
no ip address
ssid uc520-data
ssid uc520-voice
speed basic-1.0 basic-2.0 basic-5.5 6.0 9.0 basic-11.0 12.0 18.0 24.0 36.0 48.0 54.0
station-role root
interface Dot11Radio0/5/0.1
encapsulation dot1Q 1 native
bridge-group 1
bridge-group 1 subscriber-loop-control
bridge-group 1 spanning-disabled
bridge-group 1 block-unknown-source
no bridge-group 1 source-learning
no bridge-group 1 unicast-flooding
interface Dot11Radio0/5/0.100
encapsulation dot1Q 100
bridge-group 100
bridge-group 100 subscriber-loop-control
bridge-group 100 spanning-disabled
bridge-group 100 block-unknown-source
no bridge-group 100 source-learning
no bridge-group 100 unicast-flooding
interface Vlan1
no ip address
bridge-group 1
bridge-group 1 spanning-disabled
interface Vlan100
no ip address
bridge-group 100
bridge-group 100 spanning-disabled
interface BVI1
description $FW_INSIDE$
ip address 192.168.10.1 255.255.255.0
ip access-group 102 in
ip nat inside
ip virtual-reassembly in
interface BVI100
description $FW_INSIDE$
ip address 10.1.1.1 255.255.255.0
ip access-group 103 in
ip nat inside
ip virtual-reassembly in
ip forward-protocol nd
ip http server
ip http authentication local
ip http secure-server
ip http path flash:/gui
ip nat inside source list 1 interface FastEthernet0/0 overload
ip route 10.1.10.1 255.255.255.255 Integrated-Service-Engine0/0
control-plane
bridge 1 route ip
bridge 100 route ip
voice-port 0/0/0
cptone AU
voice-port 0/0/1
cptone AU
voice-port 0/0/2
cptone AU
voice-port 0/0/3
cptone AU
voice-port 0/1/0
cptone AU
voice-port 0/1/1
cptone AU
voice-port 0/2/0
translate calling 1112
connection plar opx 398
description Configured by CCA 4 FXO-0/2/0-Custom-AA
caller-id enable
voice-port 0/2/1
connection plar opx 398
description Configured by CCA 4 FXO-0/2/1-Custom-AA
caller-id enable
voice-port 0/2/2
connection plar opx 398
description Configured by CCA 4 FXO-0/2/2-Custom-AA
caller-id enable
voice-port 0/2/3
connection plar opx 398
description Configured by CCA 4 FXO-0/2/3-Custom-AA
caller-id enable
voice-port 0/4/0
auto-cut-through
signal immediate
input gain auto-control -15
description Music On Hold Port
sccp local Loopback0
sccp ccm 10.1.1.1 identifier 1 version 4.0
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate profile 1 register confprof1
dspfarm profile 1 conference
description DO NOT MODIFY, active CCA conference profile - CCA2.0 codec711
codec g711alaw
codec g711ulaw
maximum conference-participants 32
maximum sessions 2
associate application SCCP
dial-peer voice 2000 voip
description ** cue voicemail pilot number **
destination-pattern 300
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
voice-class sip outbound-proxy ipv4:10.1.10.1
dtmf-relay rtp-nte
no vad
dial-peer voice 2001 voip
description ** cue auto attendant number **
translation-profile outgoing PSTN_CallForwarding
destination-pattern 398
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
voice-class sip outbound-proxy ipv4:10.1.10.1
dtmf-relay rtp-nte
no vad
dial-peer voice 2012 voip
description ** cue prompt manager number **
translation-profile outgoing PSTN_CallForwarding
destination-pattern 739
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
voice-class sip outbound-proxy ipv4:10.1.10.1
dtmf-relay rtp-nte
no vad
dial-peer voice 90 pots
description AU-Mobile
preference 1
destination-pattern 04........
port 0/2/0
forward-digits all
no sip-register
dial-peer voice 68 pots
description NSW Number
preference 1
destination-pattern 02........
port 0/2/0
forward-digits all
no sip-register
dial-peer voice 69 pots
description TAS Number
preference 1
destination-pattern 03........
port 0/2/0
forward-digits all
no sip-register
dial-peer voice 70 pots
description WA-SA-NT number
preference 1
destination-pattern 08........
port 0/2/0
forward-digits all
no sip-register
dial-peer voice 72 pots
description QA-number
preference 1
destination-pattern 07........
port 0/2/0
forward-digits all
no sip-register
dial-peer voice 74 pots
description International number
preference 1
destination-pattern 0011T
port 0/2/0
forward-digits all
no sip-register
dial-peer voice 30 pots
description Australia-1800
preference 1
destination-pattern 1800......
port 0/2/0
forward-digits all
no sip-register
dial-peer voice 31 pots
description Australia-1300
preference 1
destination-pattern 1300......
port 0/2/0
forward-digits all
no sip-register
dial-peer voice 32 pots
description 13 Australia
preference 5
destination-pattern 13....
port 0/2/0
forward-digits all
dial-peer voice 67 pots
description mel-number
preference 1
destination-pattern 9.......
port 0/2/0
forward-digits all
no sip-register
dial-peer voice 75 pots
description mel-Number
preference 1
destination-pattern 8.......
port 0/2/0
forward-digits all
no sip-register
dial-peer voice 76 pots
description VIC number
preference 1
destination-pattern 5.......
port 0/2/0
forward-digits all
no sip-register
dial-peer voice 33 pots
description Emergency NUmber
preference 1
destination-pattern 0000
port 0/2/0
forward-digits all
no sip-register
no dial-peer outbound status-check pots
telephony-service
sdspfarm conference mute-on 111 mute-off 222
sdspfarm units 5
sdspfarm tag 1 confprof1
conference hardware
video
max-ephones 14
max-dn 56
ip source-address 10.1.1.1 port 2000
max-redirect 20
auto assign 1 to 1 type bri
calling-number initiator
service phone videoCapability 1
service phone webAccess 0
service dnis overlay
service dnis dir-lookup
timeouts interdigit 7
system message UC520
url services http://10.1.10.1/voiceview/common/login.do
url authentication http://10.1.10.1/CCMCIP/authenticate.asp
load 7906 SCCP11.9-2-1S
load 7911 SCCP11.9-2-1S
load 7931 SCCP31.9-1-1SR1S
load 7960-7940 P00308010200
load 521G-524G cp524g-8-1-17
time-zone 48
date-format dd-mm-yy
voicemail 300
max-conferences 8 gain -6
call-forward pattern .T
call-forward system redirecting-expanded
hunt-group logout HLog
multicast moh 239.10.16.16 port 2000
web admin system name cisco secret 5 $1$NPt8$6I2moMN32fQoz083VCFm90
dn-webedit
time-webedit
transfer-system full-consult dss
transfer-pattern 0.T
transfer-pattern .T
secondary-dialtone 0
night-service day Sun 17:00 09:00
night-service day Mon 17:00 09:00
night-service day Tue 17:00 09:00
night-service day Wed 17:00 09:00
night-service day Thu 17:00 09:00
night-service day Fri 17:00 09:00
night-service day Sat 17:00 09:00
create cnf-files version-stamp 7960 Dec 23 2013 10:55:20
ephone-template 15
softkeys remote-in-use Newcall
softkeys idle Redial Newcall Cfwdall Pickup Gpickup Dnd HLog Login
softkeys seized Cfwdall Endcall Redial Pickup Meetme Gpickup Callback
softkeys connected Hold Endcall Trnsfer Confrn ConfList RmLstC Acct Park Select Join
button-layout 7931 2
ephone-template 16
softkeys remote-in-use Newcall
softkeys idle Redial Newcall Cfwdall Pickup Gpickup Dnd HLog Login
softkeys seized Cfwdall Endcall Redial Pickup Meetme Gpickup Callback
softkeys connected Hold Endcall Trnsfer Confrn ConfList RmLstC Acct Park Select Join
ephone-template 17
softkeys remote-in-use CBarge Newcall
softkeys idle Redial Newcall Cfwdall Pickup Gpickup Dnd HLog Login
softkeys seized Cfwdall Endcall Redial Pickup Meetme Gpickup Callback
softkeys connected Hold Endcall Trnsfer Confrn ConfList RmLstC Acct Park Select Join
ephone-template 18
softkeys remote-in-use CBarge Newcall
softkeys idle Redial Newcall Cfwdall Pickup Gpickup Dnd HLog Login
softkeys seized Cfwdall Endcall Redial Pickup Meetme Gpickup Callback
softkeys connected Hold Endcall Trnsfer Confrn ConfList RmLstC Acct Park Select Join
button-layout 7931 2
ephone-dn 5 dual-line
number 301 no-reg primary
label 301
description PhoneA Analog
name PhoneA Analog
ephone-dn 6 dual-line
number 302 no-reg primary
label 302
description PhoneB Analog
name PhoneB Analog
ephone-dn 7 dual-line
number 303 no-reg primary
label 303
description PhoneC Analog
name PhoneC Analog
ephone-dn 8 dual-line
number 304 no-reg primary
label 304
description PhoneD Analog
name PhoneD Analog
ephone-dn 9
number BCD no-reg primary
description MoH
moh out-call ABC
ephone-dn 10 dual-line
number 201 no-reg primary
pickup-group 1
label 201
description Extension 201
name Receptionist Receptionist
mobility
call-forward busy 300
call-forward noan 300 timeout 20
ephone-dn 11 dual-line
number 207 no-reg primary
label 207
description Extension 207
name None None
ephone-dn 12 dual-line
call-waiting ring
number 203 no-reg primary
pickup-group 1
label 203
description Extension 203
name Peter Steve
call-forward busy 300
call-forward noan 300 timeout 15
huntstop channel
ephone-dn 13 dual-line
call-waiting ring
number 204 no-reg primary
pickup-group 1
label 204
description Extension 204
name Tim OConnor
call-forward busy 300
call-forward noan 300 timeout 20
huntstop channel
ephone-dn 14 dual-line
number 205 no-reg primary
pickup-group 1
label 205
description 205
name 205
ephone-dn 15 dual-line
number 206 no-reg primary
pickup-group 1
label 206
description 206
name 206
ephone-dn 16 dual-line
call-waiting ring
number 202 no-reg primary
pickup-group 1
label 202
description Extension 202
name David Holmes
call-forward busy 300
call-forward noan 300 timeout 15
huntstop channel
ephone-dn 17 dual-line
number 208 no-reg primary
label 208
description 208
name 208
ephone-dn 18 dual-line
number 209 no-reg primary
label 209
description 209
name 209
ephone-dn 19 dual-line
number 210 no-reg primary
label 210
description 210
name 210
ephone-dn 43 octo-line
number 771 no-reg primary
conference meetme
preference 3
ephone-dn 44 octo-line
number 771 no-reg primary
conference meetme
preference 2
no huntstop
ephone-dn 45 octo-line
number 771 no-reg primary
conference meetme
preference 1
no huntstop
ephone-dn 46 octo-line
number 771 no-reg primary
conference meetme
no huntstop
ephone-dn 49 octo-line
number C001 no-reg primary
conference ad-hoc
preference 3
ephone-dn 50 octo-line
number C001 no-reg primary
conference ad-hoc
preference 2
no huntstop
ephone-dn 51 octo-line
number C001 no-reg primary
conference ad-hoc
preference 1
no huntstop
ephone-dn 52 octo-line
number C001 no-reg primary
conference ad-hoc
no huntstop
ephone-dn 55
number A801... no-reg primary
mwi off
ephone-dn 56
number A800... no-reg primary
mwi on
ephone 1
device-security-mode none
mac-address 4142.4DB8.0000
ephone-template 16
max-calls-per-button 2
type anl
button 1:5
ephone 2
device-security-mode none
mac-address 4142.4DB8.0001
ephone-template 16
max-calls-per-button 2
type anl
button 1:6
ephone 3
device-security-mode none
mac-address 4142.4DB8.0002
ephone-template 16
max-calls-per-button 2
type anl
button 1:7
ephone 4
device-security-mode none
mac-address 4142.4DB8.0003
ephone-template 16
max-calls-per-button 2
type anl
button 1:8
ephone 5
device-security-mode none
mac-address 0024.97AA.E811
ephone-template 15
max-calls-per-button 2
username "Receptionist" password receptionist
type 7931
button 1:10
--More-- !
ephone 6
device-security-mode none
mac-address 0024.C4FC.4013
ephone-template 16
username "None"
type 7911
button 1:11
ephone 7
device-security-mode none
video
mac-address 000F.34FA.168B
ephone-template 16
username "steve" password petersteve
speed-dial 1 xxx label "Peter - Home"
speed-dial 2 xxx label "David - Mobile"
speed-dial 3 xxx label "Tim - Mobile AUS"
speed-dial 4 xxx label "Tim - Mobile USA"
type 7960
button 1:12
ephone 8
device-security-mode none
video
mac-address A40C.C394.B1F0
ephone-template 16
username "tim" password timoconnor
speed-dial 1 xxx label "David - Mobile"
speed-dial 2 xxx label "Peter - Mobile"
speed-dial 3 xxx label "Clare - Mobile"
type 7911
button 1:13
ephone 9
device-security-mode none
mac-address 0024.C4FC.5425
ephone-template 16
type 7911
button 1:14
ephone 10
device-security-mode none
mac-address 0024.C4FD.E27C
ephone-template 16
type 7911
button 1:15
ephone 11
device-security-mode none
video
mac-address 0007.5098.1AB6
ephone-template 16
username "holmes" password davidholmes
speed-dial 1 xx label "David - Home"
speed-dial 2 xxxl abel "Sue - Mobile"
speed-dial 3 xxx label "Peter - Mobile"
speed-dial 4 xxx label "Tim - Mobile USA"
speed-dial 5 xxx label "Tim - Mobile AUS"
type 7960
button 1:16
ephone-hunt 1 sequential
pilot 501
list 202, 203, 204
final 300
timeout 8, 8, 8
no-reg pilot
statistics collect
description Sales
alias exec cca_vm_notification schedule from_time=00 to_time=24
banner login ^CCisco Configuration Assistant. Version: 3.2 (3). Fri Nov 15 22:54:23 EST 2013^C
line con 0
no modem enable
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport input all
line vty 0 4
transport input all
line vty 5 100
transport input all
ntp master
end
UC520#I was configure custom disconnect tone refer to this site:
http://ciscoflair.blogspot.com/2009/05/cisco-fxo-disconnect-issue.html
And the tone is in the attachment, and the custom disconnect tone configuration like below:
voice class custom-cptone Disconnect
dualtone disconnect
frequency 420 420
cadence 251 255 245 250 249 250 250 250
and the port configuration was like below:
voice-port 0/1/2
supervisory disconnect dualtone mid-call
supervisory custom-cptone Disconnect
cptone NL
timeouts interdigit 4
timeouts call-disconnect 5
timeouts wait-release 5
timing hookflash-out 500
connection plar 334
impedance complex2
caller-id enable
caller-id alerting line-reversal
caller-id alerting dsp-pre-allocate
but it was not working and the phone still ringing after the PSTN caller disconnect.
but i was read about "dualtone-detect-params", and i was add the below command and i do not understand it, but it was solve hte problem:
voice class dualtone-detect-params 1
freq-max-deviation 20
cadence-variation 50
so what it is and how to determine this parameters. -
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. -
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javalenc
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