Delayed Incoming Calls

I have set up Skype Connect and connected it to my Gigaset N300IP. Outgoing calling works just fine. However, I have a problem with incoming calls. At first, I thought that incoming calling was not working at all. The Skype ID that I have associated with my SIP Profile appears as being online, and when a Skype user attempts to call that Skype ID, it simply shows as "Connecting", with no audio being returned to the caller. Then, by chance, I happened to leave a call in this state and, after 55 seconds, the call went through. After waiting the 55 seconds, the call state at the calling end changes to "Ringing" and the caller hears ringback tone. The call is detected by the N300IP, and can be answered. Once answered, audio is passed in both directions. This behaviour is repeatable -- the call will always start ringing 55 seconds after the caller initiates by pressing the call button. My N300IP is connected behind and Apple Airport Extreme configured to provide NAT. Any ideas what is causing this behaviour? Best regards, John Parker

So, a little more information. I spoke to support and was told that this is a known bug and that they are testing a fix for it, which should have been out by now. I wish Skype had a support system that provided a little more transparency into know issues. Anyway, Here is more information from our email exchange: 4/16/2013:"If the delay was as much as 45 sec then you may be seeing the attached business user bug we have had an issue for some time.,and we do have a ticket opened on that issue. That issue involves Skype version 6.1 and higher calling attached business user. The issue there is calls take 45 seconds to reach the PBX, and other times not at all." 4/17/2013:"As stated in a previous email there is an escalation open on slow connection to an attached business user from certain releases of Skype Client software. Skype Developers are testing a resolution to the problem now and hope to have it deployed before the end of the month." 4/18/2013:"The escalation number is: SIPTSOB-448.They are testing and expect to have a new release of Skype Client software ready to be released by the end of the month." 4/26/2013: "I understand you wish to know if there is a web site you can access for updates to issue SIPTSOB-448. However the ticketing system is only accessible for support staff only. You will be sent an email when updates have been made. I can also say that this issue has been escalated to highest level it can go. Also note: there is a fix they testing right now. Should be within a few weeks or so."

Similar Messages

Delay in incoming calls ringing on Lync for Windows Phone

When calling a Lync user on Windows Phone, even if the phone is connected via Wifi, the call takes about 10 seconds longer before it even starts ringing on Windows Phone. As a result it is frequently missed.
If the user is signed into the desktop client and Lync on Windows Phone, it rings immediately on the desktop client, no problem, and you can observe the delay.
Is this a known issue or limitation? Is there any way to reduce the time before the call rings on the phone?

Hi,
Did the issue also happen on IOS devices and Android Devices?
You can test with IPhone and Android Phones to check the issue.
There are several known issues for Lync mobile, you can refer to the part of “Incoming calls or instant messages (IMs) for Lync on mobile broadband” in the link below:
http://office.microsoft.com/en-001/lync-help/lync-2013-known-issues-HA102919641.aspx
Please update to the latest version Lync for WPhone and test again.
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support

Delay in softphone incoming calls

Hi All,
Softphone users are unable to hear the other party for few seconds. We are having this issue for incoming calls.
If anybody came across this issue, please let us know your suggestion
Thanks in advance !

Thanks
let me explain the issue with more info.
Scenario:
When a CIPC receives an incoming call, let say I am calling from cell phone to CIPC user, when the call get connect, for the first 4 to 5 seconds CIPC user can not hear anything, but I can hear them clearly, so there was a one way audio/delay for the first 5 seconds, after this both party able to hear. It is not occurring on all calls.
We able to duplicate this on a inbound call only, not sure whether its happening on outbound too.
Tried with Both CIPC 7.0 and 8.6, also it happens regardless how the CIPC is connected i.e., local LAN or VPN.

CME - Delay in connecting incoming calls

Hello,
For most incoming calls coming in on FXO lines we are noticing that there is a 8-10 seconds delay before the audio is established. Call comes in, IP phone rings, then there is 8-10 second delay before either party can hear anything. GW was rebooted and there has been no recent changes.
Cisco IOS Software, 2800 Software (C2800NM-ADVIPSERVICESK9-M), Version 15.1(4)M6, RELEASE SOFTWARE (fc2)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2013 by Cisco Systems, Inc.
Compiled Thu 14-Feb-13 04:14 by prod_rel_team
ROM: System Bootstrap, Version 12.3(8r)T7, RELEASE SOFTWARE (fc1)
shredit-ottawa-2853511 uptime is 2 weeks, 5 days, 22 hours, 10 minutes
System returned to ROM by reload at 17:01:24 EST Tue Feb 11 2014
System restarted at 17:03:15 EST Tue Feb 11 2014
System image file is "flash:c2800nm-advipservicesk9-mz.151-4.M6.bin"
Last reload type: Normal Reload
This product contains cryptographic features and is subject to United
States and local country laws governing import, export, transfer and
use. Delivery of Cisco cryptographic products does not imply
third-party authority to import, export, distribute or use encryption.
Importers, exporters, distributors and users are responsible for
compliance with U.S. and local country laws. By using this product you
agree to comply with applicable laws and regulations. If you are unable
to comply with U.S. and local laws, return this product immediately.
A summary of U.S. laws governing Cisco cryptographic products may be found at:
http://www.cisco.com/wwl/export/crypto/tool/stqrg.html
If you require further assistance please contact us by sending email to
[email protected].
Cisco 2821 (revision 4.0) with 473088K/51200K bytes of memory.
Processor board ID FTX0948A2RX
2 Gigabit Ethernet interfaces
1 Serial interface
2 terminal lines
1 Virtual Private Network (VPN) Module
12 Voice FXO interfaces
1 cisco service engine(s)
DRAM configuration is 64 bits wide with parity enabled.
239K bytes of non-volatile configuration memory.
250880K bytes of ATA CompactFlash (Read/Write)
License Info:
License UDI:
Device# PID SN
*0 CISCO2821 FTX0948A2RX
Configuration register is 0x2102
Please let me know if anyone has encountered this issue or has any troubleshooting steps. If more information is required let me know.
Thanks,
AK

Thanks for your reply Yosh, we do have caller-id enabled it is a service we have enabled with telco. However the phone rings correctly, once the handset is picked up, there is a 8-10 second delay befire either side can hear anything. Below is the show run requested by Chad.
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2014.03.03 16:00:52 =~=~=~=~=~=~=~=~=~=~=~=
Warning! DSPs1,2 in slot 0 are using non-default firmware from flash:
This is not recommended, the default version is 28.3.9
Current configuration : 40628 bytes
version 15.1
service tcp-keepalives-in
service timestamps debug datetime msec localtime
service timestamps log datetime msec localtime
service password-encryption
service sequence-numbers
boot-start-marker
boot system flash:c2800nm-advipservicesk9-mz.151-4.M6.bin
boot system flash:c2800nm-advipservicesk9-mz.124-3d.bin
boot-end-marker
logging buffered 500000
no logging console
enable secret 4 jHPDLZDous9V1U8vULDXrH1n.MLtRb2ON6oeuKbRZ.I
aaa new-model
aaa session-id common
memory-size iomem 10
clock timezone EST -5 0
clock summer-time EST recurring
dot11 syslog
no ip source-route
ip cef
no ip dhcp use vrf connected
ip dhcp excluded-address 10.7.3.129 10.7.3.134
ip dhcp excluded-address 10.7.3.251 10.7.3.255
ip dhcp excluded-address 10.7.3.1 10.7.3.19
ip dhcp excluded-address 10.7.3.121 10.7.3.128
ip dhcp pool voice
network 10.7.3.128 255.255.255.128
default-router 10.7.3.129
option 150 ip 10.7.3.129
lease 3
ip dhcp pool data
network 10.7.3.0 255.255.255.128
default-router 10.7.3.1
dns-server 10.2.2.10 10.2.2.11
domain-name
option 189 ascii "10.2.0.111"
lease 3
no ip domain lookup
ip domain name
no ipv6 cef
multilink bundle-name authenticated
voice-card 0
crypto pki token default removal timeout 0
interface GigabitEthernet0/0
description Trunk interface to switch - carry voice and data VLANs
no ip address
duplex auto
speed auto
interface GigabitEthernet0/0.10
description Data interface
encapsulation dot1Q 10
ip address 10.7.3.111 255.255.255.128
no ip redirects
no ip unreachables
interface GigabitEthernet0/0.20
description Voice subinterface
encapsulation dot1Q 20
ip address 10.7.3.129 255.255.255.128
no ip redirects
no ip unreachables
interface Service-Engine0/0
description Cisco Unity Express Module
ip unnumbered GigabitEthernet0/0.20
service-module ip address 10.7.3.130 255.255.255.128
service-module ip default-gateway 10.7.3.129
interface GigabitEthernet0/1
bandwidth 10000
ip address 152.192.140.94 255.255.255.252
duplex full
speed 10
service-policy output shape-etm
interface Async0/0/0
no ip address
encapsulation slip
dialer in-band
router bgp 1
bgp log-neighbor-changes
redistribute connected
neighbor 152.192.140.93 remote-as 65000
ip forward-protocol nd
ip http server
no ip http secure-server
ip http path flash:
ip route 0.0.0.0 0.0.0.0 10.100.186.109
ip route 10.7.3.130 255.255.255.255 Service-Engine0/0
tftp-server flash:CP7902060000SCCP050124A.sbin
tftp-server flash:CP7905060000SCCP050124A.sbin
tftp-server flash:CP7905060000SCCP050124A.zup
tftp-server flash:CP7912060000SCCP050124A.sbin
tftp-server flash:P00307020200.bin
tftp-server flash:P00307020200.loads
tftp-server flash:P00307020200.sb2
tftp-server flash:P00307020200.sbn
tftp-server flash:S00104000100.sbn
tftp-server flash:cmterm_7920.4.0-02-00.bin
tftp-server flash:ATA030204SCCP090202A.zup
control-plane
voice-port 0/1/0
input gain 3
echo-cancel coverage 32
cptone CA
connection plar opx 70301
description 123-1231234
caller-id enable
voice-port 0/1/1
input gain 3
echo-cancel coverage 32
cptone CA
connection plar opx 70301
description 123-1231234
caller-id enable
voice-port 0/1/2
no battery-reversal
input gain 3
echo-cancel coverage 32
cptone CA
connection plar 70301
description 123-1231234
caller-id enable
voice-port 0/1/3
input gain 3
echo-cancel coverage 32
cptone CA
connection plar opx 70301
description 123-1231234
caller-id enable
voice-port 0/2/0
input gain 3
echo-cancel coverage 32
cptone CA
connection plar opx 70301
description 123-1231234
caller-id enable
voice-port 0/2/1
input gain 3
echo-cancel coverage 32
cptone CA
connection plar opx 70301
description 123-1231234
caller-id enable
voice-port 0/2/2
input gain 3
echo-cancel coverage 32
cptone CA
connection plar opx 70301
description 123-1231234
caller-id enable
voice-port 0/2/3
input gain 3
echo-cancel coverage 32
cptone CA
connection plar opx 70301
description 123-1231234
caller-id enable
voice-port 0/3/0
input gain 3
echo-cancel coverage 32
cptone CA
connection plar opx 70353
caller-id enable
voice-port 0/3/1
input gain 3
echo-cancel coverage 32
cptone CA
connection plar opx 70353
caller-id enable
voice-port 0/3/2
input gain 3
echo-cancel coverage 32
cptone CA
connection plar opx 70353
caller-id enable
voice-port 0/3/3
input gain 3
echo-cancel coverage 32
cptone CA
connection plar opx 70353
caller-id enable
mgcp profile default
dial-peer voice 9980101 pots
preference 9
destination-pattern 9T
incoming called-number .
direct-inward-dial
port 0/1/0
dial-peer voice 9980102 pots
description Match 10-digit local calls with no delay
preference 9
destination-pattern 9[2-9].........
port 0/1/0
forward-digits 10
dial-peer voice 9980103 pots
description Match 11-digit NANP Local or LD with no delay
preference 9
destination-pattern 91[2-9].........
port 0/1/0
forward-digits 11
dial-peer voice 9980104 pots
description Match 911 emergency with no delay
preference 9
destination-pattern 9911
port 0/1/0
forward-digits 3
dial-peer voice 9980111 pots
preference 8
destination-pattern 9T
incoming called-number .
direct-inward-dial
port 0/1/1
dial-peer voice 9980112 pots
description Match 10-digit local calls with no delay
preference 8
destination-pattern 9[2-9].........
port 0/1/1
forward-digits 10
dial-peer voice 9980113 pots
description Match 11-digit NANP Local or LD with no delay
preference 8
destination-pattern 91[2-9].........
port 0/1/1
forward-digits 11
dial-peer voice 9980114 pots
description Match 911 emergency with no delay
preference 8
destination-pattern 9911
port 0/1/1
forward-digits 3
dial-peer voice 9980121 pots
preference 7
destination-pattern 9T
incoming called-number .
direct-inward-dial
port 0/1/2
dial-peer voice 9980122 pots
description Match 10-digit local calls with no delay
preference 7
destination-pattern 9[2-9].........
port 0/1/2
forward-digits 10
dial-peer voice 9980123 pots
description Match 11-digit NANP Local or LD with no delay
preference 7
destination-pattern 91[2-9].........
port 0/1/2
forward-digits 11
dial-peer voice 9980124 pots
description Match 911 emergency with no delay
preference 7
destination-pattern 9911
port 0/1/2
forward-digits 3
dial-peer voice 9980131 pots
preference 6
destination-pattern 9T
incoming called-number .
direct-inward-dial
port 0/1/3
dial-peer voice 9980132 pots
description Match 10-digit local calls with no delay
preference 6
destination-pattern 9[2-9].........
port 0/1/3
forward-digits 10
dial-peer voice 9980133 pots
description Match 11-digit NANP Local or LD with no delay
preference 6
destination-pattern 91[2-9].........
port 0/1/3
forward-digits 11
dial-peer voice 9980134 pots
description Match 911 emergency with no delay
preference 6
destination-pattern 9911
port 0/1/3
forward-digits 3
dial-peer voice 9980201 pots
preference 5
destination-pattern 9T
incoming called-number .
direct-inward-dial
port 0/2/0
dial-peer voice 9980202 pots
description Match 10-digit local calls with no delay
preference 5
destination-pattern 9[2-9].........
port 0/2/0
forward-digits 10
dial-peer voice 9980203 pots
description Match 11-digit NANP Local or LD with no delay
preference 5
destination-pattern 91[2-9].........
port 0/2/0
forward-digits 11
dial-peer voice 9980204 pots
description Match 911 emergency with no delay
preference 5
destination-pattern 9911
port 0/2/0
forward-digits 3
dial-peer voice 9980211 pots
preference 4
destination-pattern 9T
incoming called-number .
direct-inward-dial
port 0/2/1
dial-peer voice 9980212 pots
description Match 10-digit local calls with no delay
preference 4
destination-pattern 9[2-9].........
port 0/2/1
forward-digits 10
dial-peer voice 9980213 pots
description Match 11-digit NANP Local or LD with no delay
preference 4
destination-pattern 91[2-9].........
port 0/2/1
forward-digits 11
dial-peer voice 9980214 pots
description Match 911 emergency with no delay
preference 4
destination-pattern 9911
port 0/2/1
forward-digits 3
dial-peer voice 9980221 pots
preference 3
destination-pattern 9T
incoming called-number .
direct-inward-dial
port 0/2/2
dial-peer voice 9980222 pots
description Match 10-digit local calls with no delay
preference 3
destination-pattern 9[2-9].........
port 0/2/2
forward-digits 10
dial-peer voice 9980223 pots
description Match 11-digit NANP Local or LD with no delay
preference 3
destination-pattern 91[2-9].........
port 0/2/2
forward-digits 11
dial-peer voice 9980224 pots
description Match 911 emergency with no delay
preference 3
destination-pattern 9911
port 0/2/2
forward-digits 3
dial-peer voice 9980231 pots
preference 2
destination-pattern 9T
incoming called-number .
direct-inward-dial
port 0/2/3
dial-peer voice 9980232 pots
description Match 10-digit local calls with no delay
preference 2
destination-pattern 9[2-9].........
port 0/2/3
forward-digits 10
dial-peer voice 9980233 pots
description Match 11-digit NANP Local or LD with no delay
preference 2
destination-pattern 91[2-9].........
port 0/2/3
forward-digits 11
dial-peer voice 9980234 pots
description Match 911 emergency with no delay
preference 2
destination-pattern 9911
port 0/2/3
forward-digits 3
dial-peer voice 998001 voip
description CUE
destination-pattern 7030[0-7]
session protocol sipv2
session target ipv4:10.7.3.130
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 9980301 pots
preference 9
destination-pattern 9T
incoming called-number .
direct-inward-dial
port 0/3/0
dial-peer voice 9980302 pots
description Match 10-digit local calls with no delay
preference 9
destination-pattern 9[2-9].........
port 0/3/0
forward-digits 10
dial-peer voice 9980303 pots
description Match 11-digit NANP Local or LD with no delay
preference 9
destination-pattern 91[2-9].........
port 0/3/0
forward-digits 11
dial-peer voice 9980304 pots
description Match 911 emergency with no delay
preference 9
destination-pattern 9911
port 0/3/0
forward-digits 3
dial-peer voice 9980311 pots
preference 8
destination-pattern 9T
incoming called-number .
direct-inward-dial
port 0/3/1
dial-peer voice 9980312 pots
description Match 10-digit local calls with no delay
preference 8
destination-pattern 9[2-9].........
port 0/3/1
forward-digits 10
dial-peer voice 9980313 pots
description Match 11-digit NANP Local or LD with no delay
preference 8
destination-pattern 91[2-9].........
port 0/3/1
forward-digits 11
dial-peer voice 9980314 pots
description Match 911 emergency with no delay
preference 8
destination-pattern 9911
port 0/3/1
forward-digits 3
dial-peer voice 9980321 pots
preference 7
destination-pattern 9T
incoming called-number .
direct-inward-dial
port 0/3/2
dial-peer voice 9980322 pots
description Match 10-digit local calls with no delay
preference 7
destination-pattern 9[2-9].........
port 0/3/2
forward-digits 10
dial-peer voice 9980323 pots
description Match 11-digit NANP Local or LD with no delay
preference 7
destination-pattern 91[2-9].........
port 0/3/2
forward-digits 11
dial-peer voice 9980324 pots
description Match 911 emergency with no delay
preference 7
destination-pattern 9911
port 0/3/2
forward-digits 3
dial-peer voice 9980331 pots
preference 6
destination-pattern 9T
incoming called-number .
direct-inward-dial
port 0/3/3
dial-peer voice 9980332 pots
description Match 10-digit local calls with no delay
preference 6
destination-pattern 9[2-9].........
port 0/3/3
forward-digits 10
dial-peer voice 9980333 pots
description Match 11-digit NANP Local or LD with no delay
preference 6
destination-pattern 91[2-9].........
port 0/3/3
forward-digits 11
dial-peer voice 9980334 pots
description Match 911 emergency with no delay
preference 6
destination-pattern 9911
port 0/3/3
forward-digits 3
dial-peer voice 999999 pots
destination-pattern 91231231234
port 0/1/0
forward-digits 11
telephony-service
max-ephones 48
max-dn 144
ip source-address 123-1231234 port 2000
timeouts interdigit 5
load 7914 S00104000100
load 7902 CP7902060000SCCP050124A
load 7905 CP7905060000SCCP050124A
load 7910 P00403020214
load 7912 CP7912060000SCCP050124A
load 7920 cmterm_7920.3.3-01-08
load 7935 P00503010100
load 7960-7940 P00307020200
load 7970 TERM70.7-0-1-0s
load 7971 TERM70.7-0-1-0s
load ata ATA030100SCCP040211A
voicemail 70300
max-conferences 6 gain -6
call-forward pattern .T
dn-webedit
transfer-system full-consult
transfer-pattern .......
transfer-pattern ..........
transfer-pattern ...........
after-hours block pattern 1 91010 7-24
after-hours block pattern 2 91900 7-24
after-hours block pattern 5 91242 7-24
after-hours block pattern 6 91246 7-24
after-hours block pattern 7 91264 7-24
after-hours block pattern 8 91268 7-24
after-hours block pattern 9 91284 7-24
after-hours block pattern 10 91340 7-24
after-hours block pattern 11 91345 7-24
after-hours block pattern 12 91441 7-24
after-hours block pattern 13 91473 7-24
after-hours block pattern 14 91649 7-24
after-hours block pattern 15 91664 7-24
after-hours block pattern 16 91758 7-24
after-hours block pattern 17 91767 7-24
after-hours block pattern 18 91784 7-24
after-hours block pattern 19 91787 7-24
after-hours block pattern 20 91809 7-24
after-hours block pattern 21 91829 7-24
after-hours block pattern 22 91876 7-24
after-hours block pattern 23 91868 7-24
after-hours block pattern 24 91869 7-24
after-hours block pattern 25 91939 7-24
create cnf-files version-stamp 7960 Dec 21 2012 14:35:11
ephone-template 1
softkeys idle Newcall Redial Pickup Dnd Gpickup
softkeys seized Endcall Gpickup Pickup Redial
softkeys alerting Acct Callback Endcall
softkeys connected Hold Endcall Trnsfer Acct Confrn Flash
l
scheduler allocate 20000 1000
ntp update-calendar
ntp server 206.112.194.76
ntp server 206.112.202.76
end

Unity Express - Incoming calls wont get voice mail

CUE works fine with telephones on my local network. Incoming and outgoing calls work fine.
However when I get an incoming call via SIP trunk the call will not get forwarded to unity express after 10 seconds. The line goes dead.
I searched for another post which suggested the following commands:
telephony-service
call-forward pattern .T
voice service voip
allow connections from h323 to sip
I've double checked them and there's still something wrong.
Here's my current configuration:
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
h323
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
telephony-service
load 7910 P00403020214
load 7960-7940 P00305000301
max-ephones 24
max-dn 24
ip source-address 192.168.20.1 port 2000
auto assign 1 to 24
system message Comtek
voicemail 3000
max-conferences 8 gain -6
call-forward pattern .T
moh music-on-hold.au
time-webedit
transfer-system full-consult
transfer-pattern 2...
transfer-pattern 3...
directory last-name-first
directory entry 2 2001 name Phone Two 7912
directory entry 3 2000 name Phone One 7970
ephone-dn 1 dual-line
number 2000 secondary 441833000000
call-forward busy 3000
call-forward noan 3000 timeout 10
no huntstop
ephone 1
no multicast-moh
device-security-mode none
mac-address 0017.0EF0.3642
type 7970
button 1:1
So pros, any suggestions?
Thanks

I made a new dial-peer to handle incoming calls as follows.
dial-peer voice 1000 voip
description Incoming SIP
translation-profile incoming SIPin
voice-class codec 1
session protocol sipv2
incoming called-number .T
dtmf-relay rtp-nte
no vad
The translation-profile puts the call through to my 2000 extension.
This is my "show call active voice brief" when an external incoming call is ringing through to my 2000 ephone-dn.
To me this seems to show the dial-peer "1000" matching and using the g711ulaw codec
Telephony call-legs: 1
SIP call-legs: 1
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
1715 : 552 596706500ms.1 +-1 pid:1000 Answer +441833696807 connecting
dur 00:00:00 tx:0/0 rx:0/0
IP 87.127.240.98:16188 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
1715 : 553 596706510ms.1 +-1 pid:20001 Originate 2000 connecting
dur 00:00:00 tx:0/0 rx:0/0
Tele 50/0/1 (553) [50/0/1.0] tx:0/0/0ms None noise:0 acom:0 i/0:0/0 dBm
Telephony call-legs: 1
SIP call-legs: 1
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
This is the "show call active voice brief" for an external incoming call when the call is established.
Telephony call-legs: 1
SIP call-legs: 1
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
1731 : 569 597220040ms.1 +3730 pid:1000 Answer +441833696807 active
dur 00:00:02 tx:105/16800 rx:104/16640
IP 87.127.240.98:15162 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
1731 : 570 597220060ms.1 +3700 pid:20001 Originate 2000 active
dur 00:00:02 tx:0/0 rx:105/16800
Tele 50/0/1 (570) [50/0/1.0] tx:16180/16180/0ms g711ulaw noise:0 acom:0 i/0:0/0 dBm
Telephony call-legs: 1
SIP call-legs: 1
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
Not too sure where to go from here.

Analog line (FXO) Incoming calls getting connected after 3 rings

     HI,
we are having 4 Analog line (FXO)...Every time when callers call the number they hear 3 rings & after that call frwds to AA or any extension.
In show voice port summary, we can see that voice port is getting connect at the first ring but after 3 rings only phone rings.
here is the o/p of voice port.
Foreign Exchange Office 0/0/0 Slot is 0, Sub-unit is 0, Port is 0
Type of VoicePort is FXO
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Description is not set
Noise Regeneration is enabled
Non Linear Processing is enabled
Non Linear Mute is disabled
Non Linear Threshold is -21 dB
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 3 dB
Echo Cancellation is enabled
Echo Cancellation NLP mute is disabled
Echo Cancellation NLP threshold is -21 dB
Echo Cancel Coverage is set to 128 ms
Echo Cancel worst case ERL is set to 6 dB
Playout-delay Mode is set to adaptive
Playout-delay Nominal is set to 60 ms
Playout-delay Maximum is set to 1000 ms
Playout-delay Minimum mode is set to default, value 40 ms
Playout-delay Fax is set to 300 ms
Connection Mode is plar
Connection Number is 250
Initial Time Out is set to 15 s
Interdigit Time Out is set to 10 s
Call Disconnect Time Out is set to 60 s
Power Denial Disconnect Time Out is set to 1000 ms
Ringing Time Out is set to 180 s
Wait Release Time Out is set to 30 s
Companding Type is u-law
Region Tone is set for AE
Analog Info Follows:
Currently processing none
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Impedance is set to 600r Ohm
Station name None, Station number None
Caller ID Info Follows:
Standard BELLCORE
Caller ID is received after 1 ring(s)
Translation profile (Incoming): INCOMING_CallerID_PROFILE
Translation profile (Outgoing):
lpcor (Incoming):
lpcor (Outgoing):
Voice card specific Info Follows:
Signal Type is loopStart
Battery-Reversal is enabled
Number Of Rings is set to 1
Supervisory Disconnect is signal
Answer Supervision is inactive
Hook Status is On Hook
Ring Detect Status is inactive
Ring Ground Status is inactive
Tip Ground Status is inactive
Dial Out Type is dtmf
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Pulse Rate Timing is set to 10 pulses/second
InterDigit Pulse Duration Timing is set to 750 ms
Percent Break of Pulse is 65 percent
GuardOut timer is 2000 ms
Minimum ring duration timer is 125 ms
Hookflash-in Timing is set to 600 ms
Hookflash-out Timing is set to 400 ms
Supervisory Disconnect Timing (loopStart only) is set to 350 ms
OPX Ring Wait Timing is set to 6000 ms
Secondary dialtone is disabled

hostname VGUAE001
no aaa new-model
clock timezone UAE 4 0
ip cef
ip domain name yourdomain.com
no ipv6 cef
multilink bundle-name authenticated
trunk group ALL_FXO
max-retry 5
voice-class cause-code 1
hunt-scheme longest-idle
translation-profile outgoing PROFILE_ALL_FXO
voice-card 0
voice call send-alert
voice rtp send-recv
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
voice class cause-code 1
no-circuit
voice translation-rule 1112
rule 1 /^9/ //
voice translation-rule 3265
rule 1 // /9\1/
voice translation-profile INCOMING_CallerID_PROFILE
translate calling 50
voice translation-profile OUTGOING_TRANSLATION_PROFILE
translate called 1112
license udi pid CISCO2901/K9 sn FCZ173992Z8
hw-module pvdm 0/0
hw-module pvdm 0/1
username cisco privilege 15 secret 4 opjnnkXqCr4kCOa9DuALcNpBOMetBAc/usnpSWADsCI
username godiva privilege 15 secret 4 cH8b8z.ioYu/pMv/AKuEcBd/f6g9v/vm/s3aXeqUAd6
redundancy
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
ip address 192.168.31.2 255.255.255.0
ip helper-address 192.168.31.11
duplex auto
speed auto
h323-gateway voip interface
h323-gateway voip bind srcaddr 192.168.31.2
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
ip forward-protocol nd
ip http server
ip http access-class 23
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip http path flash:
ip route 0.0.0.0 0.0.0.0 192.168.31.1
control-plane
voice-port 0/0/0
trunk-group ALL_FXO 64
translation-profile incoming INCOMING_CallerID_PROFILE
groundstart auto-tip
cptone AE
connection plar opx 222
caller-id enable
voice-port 0/0/1
trunk-group ALL_FXO 64
translation-profile incoming INCOMING_CallerID_PROFILE
cptone AE
connection plar opx 222
caller-id enable
voice-port 0/0/2
trunk-group ALL_FXO 64
translation-profile incoming INCOMING_CallerID_PROFILE
cptone AE
connection plar opx 222
caller-id enable
voice-port 0/0/3
trunk-group ALL_FXO 64
translation-profile incoming INCOMING_CallerID_PROFILE
cptone AE
connection plar opx 250
caller-id enable
mgcp profile default
dial-peer voice 2000 voip
destination-pattern 2..
session target ipv4:192.168.31.11
incoming called-number .
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
dial-peer voice 10 pots
trunkgroup ALL_FXO
description **CCA*UAE*Fire**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 997
forward-digits all
no sip-register
dial-peer voice 11 pots
trunkgroup ALL_FXO
description **CCA*UAE*International Numbers**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 900T
forward-digits all
no sip-register
dial-peer voice 12 pots
trunkgroup ALL_FXO
description **CCA*UAE*Eitisalat**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 9101
forward-digits all
no sip-register
dial-peer voice 13 pots
trunkgroup ALL_FXO
description **CCA*UAE*Water or electrical emergencies**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 971
forward-digits all
no sip-register
dial-peer voice 14 pots
trunkgroup ALL_FXO
description **CCA*UAE*Police and emergencies**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 999
forward-digits all
no sip-register
dial-peer voice 15 pots
trunkgroup ALL_FXO
description **CCA*UAE*National area codes**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 9[1-579].......
forward-digits all
no sip-register
dial-peer voice 16 pots
trunkgroup ALL_FXO
description **CCA*UAE*Mobile Numbers**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 90[5-6][0-7].......
forward-digits all
no sip-register
dial-peer voice 17 pots
trunkgroup ALL_FXO
description **CCA*UAE*toll-free**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 9[2-9]00T
forward-digits all
no sip-register
dial-peer voice 18 pots
trunkgroup ALL_FXO
description **CCA*UAE*Fixed Line Numbers**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 9[2-8]T
forward-digits all
no sip-register
dial-peer voice 19 pots
trunkgroup ALL_FXO
description **CCA*UAE*808**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 9808T
forward-digits all
no sip-register
dial-peer voice 50 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/0/0
dial-peer voice 51 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/0/1
dial-peer voice 52 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/0/2
dial-peer voice 53 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/0/3
dial-peer voice 54 pots
description ** FXO pots dial-peer **
destination-pattern A0
port 0/0/0
no sip-register
dial-peer voice 55 pots
description ** FXO pots dial-peer **
destination-pattern A1
port 0/0/1
no sip-register
dial-peer voice 56 pots
description ** FXO pots dial-peer **
destination-pattern A2
port 0/0/2
no sip-register
dial-peer voice 57 pots
description ** FXO pots dial-peer **
destination-pattern A3
port 0/0/3
no sip-register
Debug vpm signal:
Nov 23 19:31:31.556: htsp_process_event: [0/0/0, FXOLS_ONHOOK, E_DSP_SIG_0000]fxols_onhook_ringing
Nov 23 19:31:31.556: htsp_timer - 125 msec
Nov 23 19:31:31.684: htsp_process_event: [0/0/0, FXOLS_WAIT_RING_MIN, E_HTSP_EVENT_TIMER]fxols_wait_ring_min_timer
Nov 23 19:31:31.684: htsp_timer - 10000 msec
Nov 23 19:31:31.684: htsp_timer3 - 5600 msec
Nov 23 19:31:31.684: [0/0/0] htsp_start_caller_id_rx:Mode BELLCORE. Alerting 0x1
Nov 23 19:31:31.684: htsp_start_caller_id_rx create dsp_stream_manager
Nov 23 19:31:31.684: [0/0/0] htsp_dsm_create_success returns 1
Nov 23 19:31:33.604: htsp_process_event: [0/0/0, FXOLS_RINGING, E_DSP_SIG_0100]
Nov 23 19:31:33.604: fxols_ringing_not
Nov 23 19:31:33.604: htsp_timer_stop
Nov 23 19:31:33.604: htsp_timer - 10000 msec
Nov 23 19:31:37.284: htsp_process_event: [0/0/0, FXOLS_RINGING, E_HTSP_EVENT_TIMER3]fxols_snoop_clid_stop
Nov 23 19:31:37.284: htsp_timer_stop3
Nov 23 19:31:37.516: htsp_process_event: [0/0/0, FXOLS_RINGING, E_DSP_SIG_0000]
Nov 23 19:31:39.604: htsp_process_event: [0/0/0, FXOLS_RINGING, E_DSP_SIG_0100]
Nov 23 19:31:39.604: fxols_ringing_not
Nov 23 19:31:39.604: htsp_timer_stop
Nov 23 19:31:39.604: htsp_timer_stop3
Nov 23 19:31:39.604: [0/0/0] htsp_stop_caller_id_rx. message length 0htsp_setup_ind
Nov 23 19:31:39.604: [0/0/0] get_fxo_caller_id:Caller ID receive failed. parseCallerIDString:no data.
Nov 23 19:31:39.604: [0/0/0] get_local_station_id calling num= calling name= calling time=11/23 23:31 orig called=
Nov 23 19:31:39.604: //-1/B583C95F8093/CCAPI/cc_api_display_ie_subfields:
   cc_api_call_setup_ind_common:
   cisco-username=
   ----- ccCallInfo IE subfields -----
   cisco-ani=
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=0
   dest=250
   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
Nov 23 19:31:39.604: //-1/B583C95F8093/CCAPI/cc_api_call_setup_ind_common:
   Interface=0x3CE27724, Call Info(
   Calling Number=,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=250(TON=Unknown, NPI=Unknown),
   Calling Translated=FALSE, Subscriber Type Str=RegularLine, FinalDestinationFlag=TRUE,
   Incoming Dial-peer=50, Progress Indication=ORIGINATING SIDE IS NON ISDN(3), Calling IE Present=FALSE,
   Source Trkgrp Route Label=ALL_FXO, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=-1
Nov 23 19:31:39.604: //-1/B583C95F8093/CCAPI/ccCheckClipClir:
   In: Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
Nov 23 19:31:39.604: //-1/B583C95F8093/CCAPI/ccCheckClipClir:
   Out: Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
Nov 23 19:31:39.604: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Nov 23 19:31:39.604: :cc_get_feature_vsa malloc success
Nov 23 19:31:39.604: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Nov 23 19:31:39.604: cc_get_feature_vsa count is 1
Nov 23 19:31:39.604: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Nov 23 19:31:39.604: :FEATURE_VSA attributes are: feature_name:0,feature_time:1025218944,feature_id:83
Nov 23 19:31:39.604: //83/B583C95F8093/CCAPI/cc_api_call_setup_ind_common:
   Set Up Event Sent;
   Call Info(Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=250(TON=Unknown, NPI=Unknown))
Nov 23 19:31:39.608: [0/0/0] htsp_dsm_close_done
Nov 23 19:31:39.608: htsp_process_event: [0/0/0, FXOLS_WAIT_SETUP_ACK, E_HTSP_SETUP_ACK]
Nov 23 19:31:39.608: fxols_wait_setup_ack:
Nov 23 19:31:39.608: [0/0/0] set signal state = 0xC timestamp = 0fxols_check_auto_call
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/cc_process_call_setup_ind:
   Event=0x22ACD828
Nov 23 19:31:39.608: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
   Try with the demoted called number 250
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCallSetContext:
   Context=0x230F9C10
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/cc_process_call_setup_ind:
   >>>>CCAPI handed cid 83 with tag 50 to app "_ManagedAppProcess_Default"
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCallProceeding:
   Progress Indication=NULL(0)
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCallSetupRequest:
   Destination=, Calling IE Present=FALSE, Mode=0,
   Outgoing Dial-peer=2000, Params=0x230FB0D0, Progress Indication=ORIGINATING SIDE IS NON ISDN(3)
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCheckClipClir:
   In: Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCheckClipClir:
   Out: Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCallSetupRequest:
   Destination Pattern=2.., Called Number=250, Digit Strip=FALSE
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCallSetupRequest:
   Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=250(TON=Unknown, NPI=Unknown),
   Redirect Number=, Display Info=
   Account Number=, Final Destination Flag=TRUE,
   Guid=B583C95F-53AC-11E3-8093-C8EEBDE4256A, Outgoing Dial-peer=2000
Nov 23 19:31:39.612: //83/B583C95F8093/CCAPI/cc_api_display_ie_subfields:
   ccCallSetupRequest:
   cisco-username=
   ----- ccCallInfo IE subfields -----
   cisco-ani=
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=0
   dest=250
   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
Nov 23 19:31:39.612: //83/B583C95F8093/CCAPI/ccIFCallSetupRequestPrivate:
   Interface=0x22847B14, Interface Type=1, Destination=, Mode=0x0,
   Call Params(Calling Number=,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=250(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
   Subscriber Type Str=RegularLine, FinalDestinationFlag=TRUE, Outgoing Dial-peer=2000, Call Count On=FALSE,
   Source Trkgrp Route Label=ALL_FXO, Target Trkgrp Route Label=, tg_label_flag=1, Application Call Id=)
Nov 23 19:31:39.612: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Nov 23 19:31:39.612: :cc_get_feature_vsa malloc success
Nov 23 19:31:39.612: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Nov 23 19:31:39.612: cc_get_feature_vsa count is 2
Nov 23 19:31:39.612: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Nov 23 19:31:39.612: :FEATURE_VSA attributes are: feature_name:0,feature_time:1025218720,feature_id:84
Nov 23 19:31:39.612: //84/B583C95F8093/CCAPI/ccIFCallSetupRequestPrivate:
   SPI Call Setup Request Is Success; Interface Type=1, FlowMode=1
Nov 23 19:31:39.612: //84/B583C95F8093/CCAPI/ccCallSetContext:
   Context=0x230FB080
Nov 23 19:31:39.612: //83/B583C95F8093/CCAPI/ccSaveDialpeerTag:
   Outgoing Dial-peer=2000
Nov 23 19:31:39.612: htsp_process_event: [0/0/0, FXOLS_PROCEEDING, E_HTSP_PROCEEDING]fxols_offhook_proc
Nov 23 19:31:39.612: htsp_timer - 120000 msec
Nov 23 19:31:39.612: //84/B583C95F8093/CCAPI/ccGetMediaClassTag:
   media class tag 0
Nov 23 19:31:39.612: //84/B583C95F8093/CCAPI/ccSetMediaclassIp2ipTags:
   media class tags set: NR 0, ASP 0
Nov 23 19:31:39.612: //83/B583C95F8093/CCAPI/ccSetMediaclassIp2ipTags:
   media class tags set: NR 0, ASP 0
Nov 23 19:31:39.612: //84/B583C95F8093/CCAPI/ccGet_xc_nr_asp_info:
   media class tags: NR 0, ASP 0
Nov 23 19:31:39.612: //83/B583C95F8093/CCAPI/ccGet_xc_nr_asp_info:
   media class tags: NR 0, ASP 0
Nov 23 19:31:39.620: //84/B583C95F8093/CCAPI/cc_api_set_called_ccm_detected:
   CallInfo(called ccm detected=TRUE ccmVersion 3)
Nov 23 19:31:39.620: //84/B583C95F8093/CCAPI/cc_api_call_proceeding:
   Interface=0x22847B14, Progress Indication=NULL(0)
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_set_called_ccm_detected:
   CallInfo(called ccm detected=TRUE ccmVersion 3)
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_set_delay_xport:
   CallInfo(delay xport=TRUE)
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_call_alert:
   Interface=0x22847B14, Progress Indication=NULL(0), Signal Indication=SIGNAL RINGBACK(1)
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_call_alert:
   Call Entry(Retry Count=0, Responsed=TRUE)
Nov 23 19:31:39.628: //83/B583C95F8093/CCAPI/ccCallAlert:
   Progress Indication=NULL(0), Signal Indication=SIGNAL RINGBACK(1)
Nov 23 19:31:39.628: //83/B583C95F8093/CCAPI/ccCallAlert:
   Call Entry(Responsed=TRUE, Alert Sent=TRUE)htsp_alert_notify
Nov 23 19:31:39.628: htsp_process_event: [0/0/0, FXOLS_PROCEEDING, E_HTSP_ALERT]fxols_offhook_alert
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_set_called_ccm_detected:
   CallInfo(called ccm detected=TRUE ccmVersion 3)
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_call_notify:
   Data Bitmask=0x5, Interface=0x22847B14, Call Id=84
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_get_ssCTreRoutingNotSupported:
   CallInfo(ssCTreRoutingNotSupported=FALSE)
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_get_ccm_detected:
   CallInfo(ccm detected=TRUE)
Nov 23 19:31:39.628: //83/B583C95F8093/CCAPI/ccCallNotify:
   Data Bitmask=0x5, Call Id=83htsp_call_service_msghtsp_call_service_msg not EFXS (2)
Nov 23 19:31:39.672: //84/B583C95F8093/CCAPI/ccIsInfoRingback:
   Returning dpRingBack=0
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/cc_api_call_connected:
   Interface=0x22847B14, Data Bitmask=0x1, Progress Indication=NULL(0),
   Connection Handle=0
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/cc_api_call_connected:
   Call Entry(Connected=TRUE, Responsed=TRUE, Retry Count=0)
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/cc_api_set_called_ccm_detected:
   CallInfo(called ccm detected=TRUE ccmVersion 3)
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/cc_api_call_notify:
   Data Bitmask=0x7, Interface=0x22847B14, Call Id=84
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/ccGenerateToneInfo:
   Stop Tone On Digit=FALSE, Tone=Null,
   Tone Direction=Network, Params=0x0, Call Id=83
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/ccConferenceCreate:
   (confID=0xFFFFFFFF, callID1=0x53, gcid=B583C95F-53AC11E3-8093C8EE-BDE4256A, tag=0x0)
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/ccConferenceCreate:
   (confID=0xFFFFFFFF, callID2=0x54, gcid=B583C95F-53AC11E3-8093C8EE-BDE4256A, tag=0x0)
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/ccConferenceCreate:
   Conference Id=0xFFFFFFFF, Call Id1=83, Call Id2=84, Tag=0x0
Nov 23 19:31:39.700: htsp_call_bridged invoked
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/cc_api_bridge_done:
   Conference Id=0x21, Source Interface=0x3CE27724, Source Call Id=83,
   Destination Call Id=84, Disposition=0x0, Tag=0xFFFFFFFF
Nov 23 19:31:39.700: //84/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
Nov 23 19:31:39.700: cc_api_get_xcode_stream : 4819
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/cc_api_bridge_done:
   Conference Id=0x21, Source Interface=0x22847B14, Source Call Id=84,
   Destination Call Id=83, Disposition=0x0, Tag=0x0
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/cc_generic_bridge_done:
   Conference Id=0x21, Source Interface=0x22847B14, Source Call Id=84,
   Destination Call Id=83, Disposition=0x0, Tag=0x0
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/ccConferenceCreate:
   Call Entry(Conference Id=0x21, Destination Call Id=84)
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/ccConferenceCreate:
   Call Entry(Conference Id=0x21, Destination Call Id=83)
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/ccConferenceCreate:
Nov 23 19:31:39.700: confID:0x21; callEntry1 callID1:0x53, type:6; callEntry2 callID2:0x54, type:1
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/cc_api_caps_ind:
   Destination Interface=0x22847B14, Destination Call Id=84, Source Call Id=83,
   Caps(Codec=0x1, Fax Rate=0x1, Fax Version:=0, Vad=0x1,
   Modem=0x2, Codec Bytes=20, Signal Type=3)
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/cc_api_caps_ind:
   Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
   Playout Max=1000(ms), Fax Nom=300(ms))
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/cc_api_get_ssCTreRoutingNotSupported:
   CallInfo(ssCTreRoutingNotSupported=FALSE)
Nov 23 19:31:39.704: //84/B583C95F8093/CCAPI/cc_api_get_ccm_detected:
   CallInfo(ccm detected=TRUE)
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/ccCallNotify:
   Data Bitmask=0x7, Call Id=83htsp_call_service_msghtsp_call_service_msg not EFXS (2)
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/cc_process_notify_bridge_done:
   Conference Id=0x21, Call Id1=83, Call Id2=84
Nov 23 19:31:39.704: //84/B583C95F8093/CCAPI/cc_api_caps_ind:
   Destination Interface=0x3CE27724, Destination Call Id=83, Source Call Id=84,
   Caps(Codec=0x1, Fax Rate=0x2, Fax Version:=0, Vad=0x1,
   Modem=0x0, Codec Bytes=160, Signal Type=2)
Nov 23 19:31:39.704: //84/B583C95F8093/CCAPI/cc_api_caps_ind:
   Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
   Playout Max=1000(ms), Fax Nom=300(ms))
Nov 23 19:31:39.704: //84/B583C95F8093/CCAPI/cc_api_caps_ack:
   Destination Interface=0x3CE27724, Destination Call Id=83, Source Call Id=84,
   Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_VOICE(0x2), Fax Version:=0, Vad=OFF(0x1),
   Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=9438)
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/cc_api_caps_ack:
   Destination Interface=0x22847B14, Destination Call Id=84, Source Call Id=83,
   Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_VOICE(0x2), Fax Version:=0, Vad=OFF(0x1),
   Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=9438)
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/ccCallConnect:
   Progress Indication=NULL(0), Data Bitmask=0x1
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/ccCallConnect:
   Call Entry(Connected=TRUE, Responsed=TRUE)
Nov 23 19:31:39.704: htsp_process_event: [0/0/0, FXOLS_PROCEEDING, E_HTSP_CONNECT]fxols_offhook_connect
Nov 23 19:31:39.704: htsp_timer_stop
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/cc_api_voice_mode_event:
   Call Id=83
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/cc_api_voice_mode_event:
   Call Entry(Context=0x230F9C10)
Nov 23 19:31:39.704: htsp_process_event: [0/0/0, FXOLS_CONNECT, E_HTSP_VOICE_CUT_THROUGH]fxols_connect_proc_voice
Nov 23 19:31:39.932: htsp_process_event: [0/0/0, FXOLS_CONNECT, E_DSP_SIG_0110]fxols_rvs_battery
Nov 23 19:31:39.932: htsp_timer_stop2
Nov 23 19:31:39.932: htsp_timer_stop2
Nov 23 19:31:48.860: //84/B583C95F8093/CCAPI/cc_api_call_disconnected:
   Cause Value=16, Interface=0x22847B14, Call Id=84
Nov 23 19:31:48.860: //84/B583C95F8093/CCAPI/cc_api_call_disconnected:
   Call Entry(Responsed=TRUE, Cause Value=16, Retry Count=0)
Nov 23 19:31:48.860: //83/B583C95F8093/CCAPI/ccConferenceDestroy:
   Conference Id=0x21, Tag=0x0
Nov 23 19:31:48.860: //83/B583C95F8093/CCAPI/ccConferenceDestroy:
Nov 23 19:31:48.860: confID:0x21; callEntry1 callID1:0x53, type:6; callEntry2 callID2:0x54, type:1
Nov 23 19:31:48.860: //83/B583C95F8093/CCAPI/cc_api_bridge_drop_done:
   Conference Id=0x21, Source Interface=0x3CE27724, Source Call Id=83,
   Destination Call Id=84, Disposition=0x0, Tag=0x0
Nov 23 19:31:48.860: //84/B583C95F8093/CCAPI/cc_api_bridge_drop_done:
   Conference Id=0x21, Source Interface=0x22847B14, Source Call Id=84,
   Destination Call Id=83, Disposition=0x0, Tag=0x0
Nov 23 19:31:48.860: //83/B583C95F8093/CCAPI/cc_generic_bridge_done:
   Conference Id=0x21, Source Interface=0x22847B14, Source Call Id=84,
   Destination Call Id=83, Disposition=0x0, Tag=0x0
Nov 23 19:31:48.864: //83/B583C95F8093/CCAPI/ccCallDisconnect:
   Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
Nov 23 19:31:48.864: //83/B583C95F8093/CCAPI/ccCallDisconnect:
   Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)
Nov 23 19:31:48.864: //83/B583C95F8093/CCAPI/cc_api_get_transfer_info:
   Transfer Number=NULL
Nov 23 19:31:48.864: //84/B583C95F8093/CCAPI/ccCallDisconnect:
   Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=16)
Nov 23 19:31:48.864: //84/B583C95F8093/CCAPI/ccCallDisconnect:
   Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)
Nov 23 19:31:48.864: //84/B583C95F8093/CCAPI/cc_api_get_transfer_info:
   Transfer Number=NULL
Nov 23 19:31:48.864: htsp_timer_stop3
Nov 23 19:31:48.872: //84/B583C95F8093/CCAPI/cc_api_get_transfer_info:
   Transfer Number=NULL
Nov 23 19:31:48.872: //84/B583C95F8093/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x22847B14, Tag=0x0, Call Id=84,
   Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)
Nov 23 19:31:48.872: //84/B583C95F8093/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
Nov 23 19:31:48.876: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Nov 23 19:31:48.876: :cc_free_feature_vsa freeing 3D1B9898
Nov 23 19:31:48.876: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Nov 23 19:31:48.876: vsacount in free is 1
Nov 23 19:31:48.884: htsp_process_event: [0/0/0, FXOLS_CONNECT, E_HTSP_RELEASE_REQ]fxols_offhook_release
Nov 23 19:31:48.884: htsp_timer_stop
Nov 23 19:31:48.884: htsp_timer_stop2
Nov 23 19:31:48.884: htsp_timer_stop3
Nov 23 19:31:48.884: [0/0/0] set signal state = 0x4 timestamp = 0
Nov 23 19:31:48.884: htsp_timer - 2000 msec
Nov 23 19:31:48.884: //83/B583C95F8093/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x3CE27724, Tag=0x0, Call Id=83,
   Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)
Nov 23 19:31:48.884: //83/B583C95F8093/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
Nov 23 19:31:48.884: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Nov 23 19:31:48.884: :cc_free_feature_vsa freeing 3D1B9978
Nov 23 19:31:48.884: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Nov 23 19:31:48.884: vsacount in free is 0
Nov 23 19:31:49.156: htsp_process_event: [0/0/0, FXOLS_GUARD_OUT, E_DSP_SIG_0110]
Nov 23 19:31:50.884: htsp_process_event: [0/0/0, FXOLS_GUARD_OUT, E_HTSP_EVENT_TIMER]fxols_guard_out_timeout
Nov 23 19:31:50.884: htsp_process_event: [0/0/0, FXOLS_ONHOOK, E_DSP_SIG_0100]

SIP incoming call with G722-64 codec not working

Hi, Guys.
Have setup cube sip trunk to ITSP, incoming and outgoing calls are working. Except for an incoming call with g722 codec and video h263 (just need voice call). The called number does not even ring. The caller informed that his using polycom phone.
Also, itsp provided 10 numbers for testing in which we can assigned to our phones but only the main number is working. When doing an incoming call, (dialing the other numbers except from the main number) can see always on the logs that itsp is always feeding the main number. I think it was because of the configuration under the sip-ua (register the maint number to a registrar) but itsp informed that it was also their setup for other clients and is working. Appreciate your help on these.
Thanks

I have looked at your logs and here are my observations..
1. When you disabled fast start on CUCM, I asked you to enable early offer on your CUBE, however I dont see this in your logs..
This is the INVITE sent to your ITSP, as you can see, this doesnt contain any SDP, that suggest you are doing delayed offer..
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.21.8.134:5060;branch=z9hG4bKC862F
From: <sip:[email protected]>;tag=AE7F464-1B0F
To: <sip:[email protected]>
Date: Thu, 03 Apr 2014 07:33:09 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,histinfo,sdp-anat
Min-SE: 1800
Cisco-Guid: 0011462194-3037647155-0083893506-2887478836
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M4
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1396510389
Contact: <sip:[email protected]:5060>
History-Info: <sip:[email protected]:5060>;index=1,<sip:[email protected]:5060>;index=2
Expires: 300
Allow-Events: telephone-event
Authorization: Digest username="AMM-4324-Trunk",realm="amcomvoice.ipsystems.com.au",uri="sip:[email protected]:5060",response="9555a4d29d9316d3f5d416f9a5096ee2",nonce="BroadWorksXhtjq88oeTdvuambBW",cnonce="6AFA84F5",qop=auth,algorithm=MD5,nc=00000001
Content-Length: 0
2. If you are doing DO, then your CUBE needs to send an answer to what your ITSP is offering in its ACK..but this is not happening
Here is what I see..Your CUBE sends SDP in its PRACK
Sent:
PRACK sip:[email protected]2.147.134.21:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.21.8.134:5060;branch=z9hG4bKC9232B
From: <sip:[email protected]>;tag=AE7F464-1B0F
To: <sip:[email protected]>;tag=SD7qfu599-1874793413-1396510390487
Date: Thu, 03 Apr 2014 07:33:09 GMT
Call-ID: [email protected]
CSeq: 103 PRACK
RAck: 323009643 102 INVITE
Allow-Events: telephone-event
Authorization: Digest username="AMM-4324-Trunk",realm="amcomvoice.ipsystems.com.au",uri="sip:[email protected]2.147.134.21:5060;transport=udp",response="a9d772d988ec971cdad556fd4a992bd0",nonce="BroadWorksXhtjq88oeTdvuambBW",cnonce="58621262",qop=auth,algorithm=MD5,nc=00000002
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 293
v=0
o=CiscoSystemsSIP-GW-UserAgent 1079 7198 IN IP4 172.21.8.134
s=SIP Call
c=IN IP4 172.21.8.134
t=0 0
m=audio 17082 RTP/AVP 8 96 100
c=IN IP4 172.21.8.134
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=ptime:20
###Here is your ACK to the 200 OK from ITSP###
On the ACK...Your CUBE doesnt include any SDP in its ACK, hence your ITSP disconnected the call immediately
ACK sip:[email protected]2.147.134.21:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.21.8.134:5060;branch=z9hG4bKCA1335
From: <sip:[email protected]>;tag=AE7F464-1B0F
To: <sip:[email protected]>;tag=SD7qfu599-1874793413-1396510390487
Date: Thu, 03 Apr 2014 07:33:09 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Authorization: Digest username="AMM-4324-Trunk",realm="amcomvoice.ipsystems.com.au",uri="sip:[email protected]:5060",response="9555a4d29d9316d3f5d416f9a5096ee2",nonce="BroadWorksXhtjq88oeTdvuambBW",cnonce="6AFA84F5",qop=auth,algorithm=MD5,nc=00000001
Allow-Events: telephone-event
Content-Length: 0
017151: Apr 3 07:33:11.089 UTC: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 202.147.134.21:5060;branch=z9hG4bKahsb3f108gbhq8pbn5k1sdj0gkbf0.1
From: <sip:[email protected]>;tag=SD7qfu599-1874793413-1396510390487
To: <sip:[email protected]>;tag=AE7F464-1B0F
Call-ID: [email protected]
CSeq: 323009054 BYE
Max-Forwards: 9
Content-Length: 0
I have two suggestions..
1. Downgrade or upgrade your CUBE IOS. Something is not quite right with this behaviour
2. Send your full sh run
On your inbound call issue, you need to send me the logs for a call to another of your DDI..

An incoming call from VoIP. The SPA122 generate Dial Tone after the far end hung up rather busy tone.

Hi All!
I have a problem with the SPA122 telephony adapter, uncorrectly process the subscriber signaling at the end of the call.
1) Outbound call from FXS port SPA122 . When a remote caller hangs up first , the subscriber SPA122 Reorder Tone played with a delay specified in the Reorder Delay. This circuit is working properly.
2 ) Incoming call from VoIP to the SPA122. When a remote caller hangs up first , subscriber on the FXS port of the SPA122 hears silence ~ 3-4 seconds , then SPA122 plays Dial Tone, as if he had just picked up the phone and he 's going to call . No signal lights out (Busy Tone or Reorder Tone) will not play .
Config is attached.
Model: SPA122, LAN, 2 FXS
Hardware Version: 1.0.0 Boot Version: 1.0.1 (Oct 6 2011 - 20:04:00)
Firmware Version: 1.3.2-XU (014) Jul 2 2013
Recovery Firmware: 1.0.2 (001)
WAN MAC Address: 6C:20:56:55:3A:B6
Host Name: SPA122
Domain Name: (none)
Serial Number: CCQ16450LG3
However, other VoIP terminals registered to Huawei, including older versions of the Linksys SPA2102 work in these scenarios correctly.
Where to kick it?

[2] is misconfiguration on your's side. You have CPC turned on, but no CPC capable device. Set CPC Duration to zero to turn off CPC.
By the way, wrong forum for your question. You should consider to move it to space.

Just Updated iPhone 3GS to iOS5, Can't receive any incoming calls at all

Hi, I just updated my phone to iOS5 and now, I'm having these troubles.
- Can't receive any incoming calls at all
- The incoming calls directly goes to voice mail greeting
- No incoming rings & no missed calls alerts
- Also noticed the incoming text with 4 to 5 hours delays
- Sometimes 3G network get lost
The reception (status bar) is good level, can make outgoing calls okay and can send text okay.
I did try reset the network setting and it seems working a few hours then went back to troubles (mentioned above).
I've done 3 or 4 times reset but none have been resolved yet.
Anyone having a same issues?
I have
iPhone 3GS with iOS5
Network with O2 UK
Thanks,

After reset "All network settings" a several times, it seems working but few hours later it 's been gone back to troubles again.
Is this related to iOS5 or O2?
I've contacted the network carrier several times but no-one yet get back to me.
Any ideas, please helps.
Re: iOS5 updated and all incoming calls going straight to voicemail

Nomorobo blocking all incoming calls

I set up Nomorobo last night. Got the confirmation phone call. After that, all incoming calls were blocked after the first ring. I emalied Nomorobo and am waiting for their response. Anyone else have this experience?

csk47 wrote:
I just installed Nomorobo yesterday as per the instructions given. Everything appeared to be set up correctly and the test screen showed protected. A couple of hours later I got a call with a short first ring then a second ring..then nothing. So I assumed it was a robocall that was intercepted. A few hours later another call came in..it, too, was intercepted. It seemed odd that two calls coming in..both are robocalls..so I logged into Xfinity and saw that both calls went right to voice mail rather than ringing through so I could answer them at home. Tried calling my home number with my cell and same thing...right to voice mail. Sent a message to Nomorobo but no response as yet. It seems the fault must be in the Call Forwarding preferences of Xfinity...but for now I've disconnected Nomorobo and calls from my cell go correctly to my home phone. If anyone comes up with a solution please post it.It took a month for me to hear back from nomorobo support and then I got this when I told them all my calls were going to robo. I did have it set up correctly so still not sure why mine won't work and I undid everything so I am back to 'normal' for now. It might be of some use to you.... "I'm sincerely sorry for the extreme delay here. From the information you've given me it sounds like the toll-free number (202-813-1600) we provided you is in your Call Forwarding list, not the Advanced Call Forwarding list, which means all your calls are being forwarded to Nomorobo. You should remove the number from that location and all incoming calls will no longer be blocked.If you don't see the Advanced Call Forwarding option, it's because Voice2Go isn't activated. Here is a link that should help you activate Voice2Go: https://customer.comcast.com/help-and-support/phone/activate-voice-2go . Here is another link with specific instructions to using Nomorobo on Comcast Xfinity: https://customer.comcast.com/help-and-support/phone/nomorobo"

IPhone Voice Memos Interrupted By Incoming Calls

I notice that every time I use my Voice Memos software on the 3G(s), if an call comes in, the Voice Memo stops recording and I won't know it till I look at the phone later. Is there ANY way to stop that for that not to happen and you can have perhaps the calls go directly into voice mail so that the Voice Memo feature doesn't get interrupted? It's useless to me otherwise.
Thanks for anyone's input!

The voice recorder application - just like every other iPhone application, native or 3rd party - will shut down when you receive a phone call. The iPhone operates under the assumption that the device is first and foremost a telephone.
As far as work arounds go, you can always put the phone in airplane mode when you need to use the voice recorder app uninterrupted. This will send all incoming calls directly to voicemail.
Message was edited by: Ansuz82

I can call out but all incoming calls are going directly to VM with no signal that I missed a call or have VM. I can send/receive text messages from other iPhone/iPad/iTouch users, I am assuming this is really "iMessage." Is it apple software?

I have spent hours on the phone with Apple and AT&T. I went thought reset, on/off, etc, etc with both companies to no avail. Each company pointed fingers at the other....as being the source of the problem.
Problems: Suddenly ALL incoming calls were going directly to VM with no signal I missed calls and/or had VM. I was also unable to receive all Text Messages...Oddly, I could send text messages to anyone (even non-apple users but I could not receive their responses)........then I when I got home I started receiving text messages from other apples users ONLY. I assume now - iMessage kicked in and I could text (send/receive) other iPhone/iPad/iTouch users ONLY. ....yes, I could still (send) text messages to my husband's blackberry (he received my messages fine) but my phone would NOT receive his text respones.
Finally, I googled the problem and found this community where other people have had the exact same problems! One person said he "turned off 3 G" which was the solution for him....so I did the same and VIOLA! My problem solved! Nevermind the fact that I pay for 3G and cannot use it....so here's my question, if 3G is the problem on my phone is this an APPLE issue or a NETWORK problem? Do I purchase a new phone and slip in my same SIM card and hope the same does not occur or do I get a whole new SIM card and phone? What is the long term resolution to this problem?
I am happy however to find that my problem is NOT an isolated incident and wish Apple or AT&T had told me this is not so uncommon because I thought (based on the baffled response from Apple) that this has never occurred before. Where is Steve Jobs when we need him?

jsavage9621,
It pains me to hear about your experience with the Home Phone Connect. This device usually works seamlessly and is a great alternative to a landline phone. It sounds like we've done our fair share of work on your account here. I'm going to go ahead and send you a Private Message so that we can access your account and review any open tickets for you. I look forward to speaking with you.
TrevorC_VZW
Follow us on Twitter @VZWSupport

I am facing issue in Receiving incoming calls, Name not getting displayed though the same has been saved in my phone book!! I have done sync from Windows contacts.. please help if some1 knows how to rectify the issue...

I am facing issue in Receiving incoming calls, Name not getting displayed though the same has been saved in my phone book!! I have done sync from Windows contacts.. please help if some1 knows how to rectify the issue...

Has your carrier been having issues with Call Display? Do the telephone numbers come up when people call, or does it just show 'Unknown Number' or 'Blocked' ?

IPhone 5s is not receiving incoming calls

Hi everyone,
I am having problems with receiving incoming calls after I got the 5s. Sometimes, it didn't even ring, just went straight to voice mail even though the signal was strong. Sometimes they have to call me up to 8 - 10 times for the phone to ring once and the other calls will appear as missed calls. Does anyone experience the same problem with the 5s? Should I bring it to the apple store for them to take a look?
Thanks

Double check settings - do not disturb - off

Iphone 5s cannot call out any calls, but still can receive incoming calls and sms

Hi
anyone experiencing this problem? After buying iphone 5s less than 2 months which I never did in the past, I could no longer make any phone calls out, yet i still can receive incoming calls and data.
Could someone kindly assist in this?

Contact your carrier. Phone functions are a carrier responsibility. Have them check your account. Are you seeing any error messages?

Delayed Incoming Calls

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