Incoming calls issue with 8800 carbon
Hi I just bought an 8800 carbon but I'm having an answering issue. I can make calls absolutely fine but when receiving calls, it resets my phone (the caller gets a busy tone and the phone goes back to the intro screen & tries to get service again)... No idea what the problem could be. I'm with Fido in Canada. Any suggestions?
Thanks
hey guys I found the problem. It was the name display option with Fido. I removed it from my plan and its all good now!
Similar Messages
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Calling issue with Cisco 7937 conference station
Hi Friends,
I am facing issue wiht Cisco 7937 conference station, our customer have various branch offices accross the world. All branches are connected over MPLS through service provider( SIP service provider) . there is a centralized CUCM and remote office have SIP Voice gateways .
When making calls from once remote site to another using Cisco 6921 phones calls working fine
When making calls from once remote site to another using Cisco 7937 conference station to make call any phone at remote office, calls are getting disconneted, remote phone rings when calls, but its gets fast busy tone when other party picks up the phone and not able to talk.
I suspect the issue with Codec but we have configured transcoders in VG and registered with CUCM
Please help me if any one experience such issue earlier.
Regards
Sivahi Basant,
1. Actually tow phones A and B are registerd with centralized CUCM, A and B are located in two different locations, RTP traffic between And B pass through service provider.
Call Flow --> Phone A ---->CUCMRouterpattern--> SIP trunk ----> Voice gateway--->Service provider cloud---> Respective Voice Gateway---> CUCM -- Phone B
Show Run
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2014.02.27 15:14:52 =~=~=~=~=~=~=~=~=~=~=~=
sh run
Building configuration...
Current configuration : 12139 bytes
! Last configuration change at 06:35:59 UTC Tue Feb 25 2014
! NVRAM config last updated at 11:16:38 UTC Mon Feb 24 2014 by administrator
! NVRAM config last updated at 11:16:38 UTC Mon Feb 24 2014 by administrator
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname eucamvgw01
boot-start-marker
boot system flash:c2900-universalk9-mz.SPA.151-4.M5.bin
boot-end-marker
card type e1 0 0
logging buffered 51200 warnings
no logging console
no aaa new-model
no network-clock-participate wic 0
no ipv6 cef
ip source-route
ip traffic-export profile cuecapture mode capture
bidirectional
ip cef
ip multicast-routing
ip domain name drreddys.eu
ip name-server 10.197.20.1
ip name-server 10.197.20.2
multilink bundle-name authenticated
stcapp ccm-group 2
stcapp
stcapp feature access-code
stcapp feature speed-dial
stcapp supplementary-services
port 0/1/0
fallback-dn 5428025
port 0/1/1
fallback-dn 5428008
port 0/1/2
fallback-dn 5421462
port 0/1/3
fallback-dn 5421463
isdn switch-type primary-net5
crypto pki token default removal timeout 0
voice-card 0
dsp services dspfarm
voice call send-alert
voice call disc-pi-off
voice call convert-discpi-to-prog
voice rtp send-recv
voice service voip
ip address trusted list
ipv4 10.198.0.0 255.255.255.0
ipv4 152.63.1.0 255.255.255.0
address-hiding
allow-connections sip to sip
no supplementary-service h225-notify cid-update
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
fax-relay ans-disable
sip
rel1xx supported "track"
privacy pstn
no update-callerid
early-offer forced
call-route p-called-party-id
voice class uri 100 sip
host 41.206.187.71
voice class codec 10
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 ilbc
codec preference 4 g729r8
codec preference 5 g729br8
voice class codec 20
codec preference 1 g729br8
codec preference 2 g729r8
voice moh-group 1
moh flash:moh/Panjo.alaw.wav
description MOH G711 alaw
multicast moh 239.1.1.2 port 16384 route 10.198.2.9
voice translation-rule 1
rule 1 /^012237280\(..\)/ /54280\1/
rule 2 /^012236514\(..\)/ /54214\1/
rule 3 /^01223651081/ /5428010/
rule 4 /^01223506701/ /5428010/
voice translation-rule 2
rule 1 /^00\(.+\)/ /+\1/
rule 2 /^0\(.+\)/ /+44\1/
rule 3 /^\([0-9].+\)/ /+\1/
voice translation-rule 3
rule 1 /^9\(.+\)/ /\1/
rule 2 /^\+44\(.+\)/ /0\1/
rule 3 /^\+\(.+\)/ /00\1/
voice translation-rule 4
rule 1 /^54280\(..\)/ /12237280\1/
rule 2 /^54214\(..\)/ /12236514\1/
rule 3 /^\+44\(.+\)/ /\1/
rule 4 /^.54280\(..\)/ /12237280\1/
rule 5 /^.54214\(..\)/ /12236514\1/
voice translation-rule 9
rule 1 /^\(....\)/ /542\1/
voice translation-rule 10
voice translation-rule 11
rule 1 /^\+44122372\(....\)/ /542\1/
rule 2 /^\+44122365\(....\)/ /542\1/
voice translation-rule 12
voice translation-rule 13
rule 1 /^\([18]...\)/ /542\1/
voice translation-rule 14
voice translation-profile MPLS-incoming
translate calling 10
translate called 9
voice translation-profile MPLS-outgoing
translate calling 11
translate called 12
voice translation-profile PSTN-incoming
translate calling 2
translate called 1
voice translation-profile PSTN-outgoing
translate calling 4
translate called 3
voice translation-profile SRST-incoming
translate calling 14
translate called 13
license udi pid CISCO2921/K9 sn FGL145110RE
hw-module ism 0
hw-module pvdm 0/0
username administrator privilege 15 secret 5 $1$syu5$DsxdOgfS7Wltx78o4PV.60
redundancy
controller E1 0/0/0
ip tcp path-mtu-discovery
ip scp server enable
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
description internal LAN
ip address 10.198.2.9 255.255.255.0
duplex auto
speed auto
interface ISM0/0
ip unnumbered GigabitEthernet0/0
service-module ip address 10.198.2.8 255.255.255.0
!Application: CUE Running on ISM
service-module ip default-gateway 10.198.2.9
interface GigabitEthernet0/1
description to TATA NGN
ip address 115.114.225.122 255.255.255.252
duplex auto
speed auto
interface GigabitEthernet0/2
description SIP Trunks external
ip address 79.121.254.83 255.255.255.248
ip access-group SIP-InBound in
ip traffic-export apply cuecapture size 8000000
duplex auto
speed auto
interface ISM0/1
description Internal switch interface connected to Internal Service Module
no ip address
shutdown
interface Vlan1
no ip address
ip forward-protocol nd
no ip http server
no ip http secure-server
ip route 0.0.0.0 0.0.0.0 10.198.2.1
ip route 10.198.2.8 255.255.255.255 ISM0/0
ip route 41.206.187.0 255.255.255.0 115.114.225.121
ip route 77.37.25.46 255.255.255.255 79.121.254.81
ip route 83.245.6.81 255.255.255.255 79.121.254.81
ip route 83.245.6.82 255.255.255.255 79.121.254.81
ip route 95.223.1.107 255.255.255.255 79.121.254.81
ip route 192.54.47.0 255.255.255.0 79.121.254.81
ip access-list extended SIP-InBound
permit ip host 77.37.25.46 any
permit ip host 83.245.6.81 any
permit ip host 83.245.6.82 any
permit ip 192.54.47.0 0.0.0.255 any
permit icmp any any
permit ip host 95.223.1.107 any
deny ip any any log
control-plane
voice-port 0/1/0
compand-type a-law
timeouts initial 60
timeouts interdigit 60
timeouts ringing infinity
caller-id enable
voice-port 0/1/1
compand-type a-law
timeouts initial 60
timeouts interdigit 60
timeouts ringing infinity
caller-id enable
voice-port 0/1/2
compand-type a-law
timeouts initial 60
timeouts interdigit 60
timeouts ringing infinity
caller-id enable
voice-port 0/1/3
compand-type a-law
timeouts initial 60
timeouts interdigit 60
timeouts ringing infinity
caller-id enable
no ccm-manager fax protocol cisco
ccm-manager music-on-hold bind GigabitEthernet0/0
ccm-manager config server 152.63.1.19 152.63.1.100 172.27.210.5
ccm-manager sccp local GigabitEthernet0/0
ccm-manager sccp
mgcp profile default
sccp local GigabitEthernet0/0
sccp ccm 10.198.2.9 identifier 3 priority 3 version 7.0
sccp ccm 152.63.1.19 identifier 4 version 7.0
sccp ccm 152.63.1.100 identifier 5 version 7.0
sccp ccm 172.27.210.5 identifier 6 version 7.0
sccp
sccp ccm group 2
bind interface GigabitEthernet0/0
associate ccm 4 priority 1
associate ccm 5 priority 2
associate ccm 6 priority 3
associate ccm 3 priority 4
associate profile 1002 register CFB_UK_CAM_02
associate profile 1001 register XCODE_UK_CAM_02
associate profile 1000 register MTP_UK_CAM_02
dspfarm profile 1001 transcode
codec ilbc
codec g722-64
codec g729br8
codec g729r8
codec gsmamr-nb
codec pass-through
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 18
associate application SCCP
dspfarm profile 1002 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 2
associate application SCCP
dspfarm profile 1000 mtp
codec g711alaw
maximum sessions software 200
associate application SCCP
dial-peer cor custom
name SRSTMode
dial-peer cor list SRST
member SRSTMode
dial-peer voice 100 voip
description *** Inbound CUCM ***
translation-profile incoming PSTN-incoming
incoming called-number .
voice-class codec 10
voice-class sip call-route p-called-party-id
dtmf-relay rtp-nte
no vad
dial-peer voice 500 voip
description *** Inbound TATA MPLS ***
translation-profile incoming MPLS-incoming
session protocol sipv2
session target sip-server
incoming called-number ....
incoming uri from 100
voice-class codec 20
dtmf-relay rtp-nte
no vad
dial-peer voice 510 voip
description *** Outbound TATA MPLS ***
translation-profile outgoing MPLS-outgoing
destination-pattern 54[013-9]....
session protocol sipv2
session target ipv4:41.206.187.71
session transport udp
voice-class codec 20
dtmf-relay rtp-nte
no vad
dial-peer voice 520 voip
description *** Outbound TATA MPLS ***
translation-profile outgoing MPLS-outgoing
destination-pattern 5[0-35-9].....
session protocol sipv2
session target ipv4:41.206.187.71
session transport udp
voice-class codec 20
dtmf-relay rtp-nte
no vad
dial-peer voice 200 voip
description *** Inbound M12 *** 01223651081, 01223651440 - 01223651489
translation-profile incoming PSTN-incoming
session protocol sipv2
session target sip-server
session transport udp
incoming called-number 0122365....
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 201 voip
description *** Inbound M12 *** 012237280XX
translation-profile incoming PSTN-incoming
session protocol sipv2
session target sip-server
session transport udp
incoming called-number 012237280..
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 202 voip
description *** Inbound M12 *** 01223506701
translation-profile incoming PSTN-incoming
session protocol sipv2
session target sip-server
session transport udp
incoming called-number 01223506701
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 210 voip
description *** Outbound M12 ***
translation-profile outgoing PSTN-outgoing
destination-pattern +...T
session protocol sipv2
session target ipv4:83.245.6.81
session transport udp
dtmf-relay rtp-nte
codec g711alaw
no vad
dial-peer voice 211 voip
description *** Outbound ISDN for SRST and emergency ***
translation-profile outgoing PSTN-outgoing
destination-pattern 9.T
session protocol sipv2
session target ipv4:83.245.6.81
session transport udp
dtmf-relay rtp-nte
codec g711alaw
no vad
dial-peer voice 212 voip
description *** Outbound ISDN for emergency ***
translation-profile outgoing PSTN-outgoing
destination-pattern 11[02]
session protocol sipv2
session target ipv4:83.245.6.81
session transport udp
dtmf-relay rtp-nte
codec g711alaw
no vad
dial-peer voice 2000 voip
description *** Outbound to CUCM Primary ***
preference 1
destination-pattern 542....
session protocol sipv2
session target ipv4:152.63.1.19
voice-class codec 10
voice-class sip call-route p-called-party-id
dtmf-relay rtp-nte
no vad
dial-peer voice 2001 voip
description *** Outbound to CUCM Secondary ***
preference 2
destination-pattern 542....
session protocol sipv2
session target ipv4:152.63.1.100
voice-class codec 10
voice-class sip call-route p-called-party-id
dtmf-relay rtp-nte
no vad
dial-peer voice 2002 voip
description *** Outbound to CUCM Teritiary ***
preference 3
destination-pattern 542....
session protocol sipv2
session target ipv4:172.27.210.5
voice-class codec 10
voice-class sip call-route p-called-party-id
dtmf-relay rtp-nte
no vad
dial-peer voice 999010 pots
service stcapp
port 0/1/0
dial-peer voice 999011 pots
service stcapp
port 0/1/1
dial-peer voice 999012 pots
service stcapp
port 0/1/2
dial-peer voice 999013 pots
service stcapp
port 0/1/3
sip-ua
no remote-party-id
gatekeeper
shutdown
call-manager-fallback
secondary-dialtone 9
max-conferences 4 gain -6
transfer-system full-consult
ip source-address 10.198.2.9 port 2000
max-ephones 110
max-dn 400 dual-line no-reg
translation-profile incoming SRST-incoming
moh flash:/moh/Panjo.ulaw.wav
multicast moh 239.1.1.1 port 16384 route 10.198.2.9
time-zone 22
time-format 24
date-format dd-mm-yy
line con 0
login local
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line 131
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0 4
session-timeout 60
exec-timeout 60 0
privilege level 15
login local
transport input all
line vty 5 15
session-timeout 60
exec-timeout 60 0
privilege level 15
login local
transport input all
scheduler allocate 20000 1000
ntp server 10.1.30.1
end
eucamvgw01#
Sh SCCP
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2014.03.03 17:57:44 =~=~=~=~=~=~=~=~=~=~=~=
SCCP Admin State: UP
Gateway Local Interface: GigabitEthernet0/0
IPv4 Address: 10.198.2.9
Port Number: 2000
IP Precedence: 5
User Masked Codec list: None
Call Manager: 10.198.2.9, Port Number: 2000
Priority: 3, Version: 7.0, Identifier: 3
Call Manager: 152.63.1.19, Port Number: 2000
Priority: N/A, Version: 7.0, Identifier: 4
Trustpoint: N/A
Call Manager: 152.63.1.100, Port Number: 2000
Priority: N/A, Version: 7.0, Identifier: 5
Trustpoint: N/A
Call Manager: 172.27.210.5, Port Number: 2000
Priority: N/A, Version: 7.0, Identifier: 6
Trustpoint: N/A
MTP Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 1000
Reported Max Streams: 400, Reported Max OOS Streams: 0
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
TLS : ENABLED
Transcoding Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 1001
Reported Max Streams: 36, Reported Max OOS Streams: 0
Supported Codec: ilbc, Maximum Packetization Period: 120
Supported Codec: g722r64, Maximum Packetization Period: 30
Supported Codec: g729br8, Maximum Packetization Period: 60
Supported Codec: g729r8, Maximum Packetization Period: 60
Supported Codec: gsmamr-nb, Maximum Packetization Period: 60
Supported Codec: pass-thru, Maximum Packetization Period: N/A
Supported Codec: g711ulaw, Maximum Packetization Period: 30
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: g729ar8, Maximum Packetization Period: 60
Supported Codec: g729abr8, Maximum Packetization Period: 60
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
Conferencing Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 1002
Reported Max Streams: 16, Reported Max OOS Streams: 0
Supported Codec: g711ulaw, Maximum Packetization Period: 30
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: g729ar8, Maximum Packetization Period: 60
Supported Codec: g729abr8, Maximum Packetization Period: 60
Supported Codec: g729r8, Maximum Packetization Period: 60
Supported Codec: g729br8, Maximum Packetization Period: 60
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
TLS : ENABLED
Alg_Phone Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Device Name: AN71FEF7F070080
Reported Max Streams: 1, Reported Max OOS Streams: 0
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: g711ulaw, Maximum Packetization Period: 20
Supported Codec: g711alaw, Maximum Packetization Period: 20
Supported Codec: g729r8, Maximum Packetization Period: 220Supported Codec: g729ar8, Maximum Packetization Period: 220
Supported Codec: g729br8, Maximum Packetization Period: 220
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: ilbc, Maximum Packetization Period: 120
Alg_Phone Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Device Name: AN71FEF7F070081
Reported Max Streams: 1, Reported Max OOS Streams: 0
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: g711ulaw, Maximum Packetization Period: 20
Supported Codec: g711alaw, Maximum Packetization Period: 20
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: g729ar8, Maximum Packetization Period: 220
Supported Codec: g729br8, Maximum Packetization Period: 220
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: ilbc, Maximum Packetization Period: 120
Alg_Phone Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Device Name: AN71FEF7F070082
Reported Max Streams: 1, Reported Max OOS Streams: 0
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: g711ulaw, Maximum Packetization Period: 20Supported Codec: g711alaw, Maximum Packetization Period: 20
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: g729ar8, Maximum Packetization Period: 220
Supported Codec: g729br8, Maximum Packetization Period: 220
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: ilbc, Maximum Packetization Period: 120
Alg_Phone Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Device Name: AN71FEF7F070083
Reported Max Streams: 1, Reported Max OOS Streams: 0
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: g711ulaw, Maximum Packetization Period: 20
Supported Codec: g711alaw, Maximum Packetization Period: 20
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: g729ar8, Maximum Packetization Period: 220
Supported Codec: g729br8, Maximum Packetization Period: 220
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: ilbc, Maximum Packetization Period: 120
eucamvgw01# -
Incoming calls issue in Third Party SIP Phone
Hi,
Yesterday I configured my third party sip phone which is yealink in this case on cucm and successfully registered it with cucm, despite of registration i have some calling issue in this phone. I am able to make outbound calls from this phone to any other phone however issue is related to inbound calls.I tried calling its DN from anywhere but call disconnect after sometime. Also didnt get any proper sip session trace in RTMT. Kindly suggest some step to sortout this issue.
ThanksDear Manish,
Call normally dicsonnected after 30-40 sec with termination code 102 in session trace. PFB SDI trace with 5030 is Thirdparty sip phone and 5033 is c7945. Looking forward for your suggestion.
CallingPartyNumber=5033
|DialingPartition=
|DialingPattern=5030
|FullyQualifiedCalledPartyNumber=5030
|DialingPatternRegularExpression=(5030)
|DialingWhere=
|PatternType=Enterprise
|PotentialMatches=NoPotentialMatchesExist
|DialingSdlProcessId=(0,0,0)
|PretransformDigitString=5030
|PretransformTagsList=SUBSCRIBER
|PretransformPositionalMatchList=5030
|CollectedDigits=5030
|UnconsumedDigits=
|TagsList=SUBSCRIBER
|PositionalMatchList=5030
|VoiceMailbox=
|VoiceMailCallingSearchSpace=PT-LHR-LOCAL:PT-Local:Unityvmpt:PT-F6-Local:PT-ISL-LOCAL:PT-KHI-LOCAL:PT_Operator_LHR:PT_Operator_KHI:PT_Operator_ISL
|VoiceMailPilotNumber=7103
|RouteBlockFlag=RouteThisPattern
|RouteBlockCause=0
|AlertingName=Syed Ahmer
|UnicodeDisplayName=Syed Ahmer
|DisplayNameLocale=1
|OverlapSendingFlagEnabled=0
12:17:38.028 |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 172.16.200.21:[5062]:
[23928282,NET]
INVITE sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.11:5060;branch=z9hG4bK1ca0cc6e317649
From: "Syed Ahmer" ;tag=8787406~039e2a80-8561-4586-8954-d01ed2aa12c8-246211918
To:
Date: Thu, 30 Jan 2014 07:17:38 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Send-Info: conference, x-cisco-conference
Alert-Info:
Contact:
Remote-Party-ID: "Syed Ahmer" ;party=calling;screen=yes;privacy=off
Max-Forwards: 70
Content-Length: 0
|14,100,50,1.14103336^10.163.14.4^SEP00230432C828
12:17:38.028 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 0|14,100,50,1.14103336^10.163.14.4^SEP00230432C828
12:17:38.028 |EnvProcessUdpHandler::fireSignal - SEND: index = 0, handler = 0xaf299320|*^*^*
12:17:38.028 |EnvProcessUdpPort::fireSignal - SEND, destination = 172.16.200.21:5062|*^*^*
12:17:38.028 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 850, 172.16.200.21:5062)|*^*^* -
Video group call issue with chat
when i'm on a one-on-one call with someone, i can use my video and chat with them just fine
but when they add me to group calls, i cant turn on my camera and i can't send typed messages in the chat window
i have the most recent version of skype, i've even uninstalled and reinstalled and it still wont workIf there are more than 10 persons in a chat, Video will be unavailable. This is aknown bug and will be fixed in the next incremental version of Skype.
For text, type /get name in the groupchat, if it returns #skypename your in an old p2p chat that your client might have issues with, if the return is 19: , your in a new cloudchat, and in theory all should be good.
Post 6.21 clients have issues with p2p chats, and pre have issues with cloud ones. I'm including the 6.21 installer link so you may downgrade if you find it necessary:
http://download.skype.com/msi/SkypeSetup_6.21.0.104.msi -
Incoming messages issues with Mail application
I'm using an email account attached to my internet company and set up with the "Mail Application" on my Mac. For the past week, When I turn "Mail" on, the Mail Activity window would show me the incoming message in progress, ex: 1 out of 26. The process would move forward showing also the speed of the reception (KB/s) and for no reason it would stop in the middle and stop receiving any email. It would basically freeze and I have to quit Mail by "force quit". If I re-open it again, it would be doing the exact same thing and receiving again the first couple emails, always the same one that I already received and freeze in the middle. I basically can't use Mail at all lately. Any idea to fix this problem?
ThanksHi there,
cool, I'm not alone...
I can't remember exactly, when the problem started. I guess it was around the 10.5.2 or 10.5.3 update. Before (also in Leopard) Mail worked fine.
Now I have to do the following procedure to receive an email (note: I have to repeat this for each email):
1. Check, if mail is receiving an email (because it never completes receiving on it's own, I'm never notified of the new email).
2. If Mail is receiving an email (check the Activity window), I have to force quit mail
3. Launch Mail again, immediately hurrying to the Mailbox menu and setting all accounts to offline.
By now, the email has already been received from the server but not internally marked as such, so staying online would result in fetching the same email again – hanging 'afterwards' included.
4. Select the Inbox and open the Get Account Info window by right clicking on the Inbox or pressing cmd + i.
5. Select the corresponding account from the list (I got three POP3 accounts), e.g. the one for which the last email has been received.
6. Manually remove this email (and only this one) from the server.
7. Close the Get Account Info window and take all accounts online again.
8. Check for new mail...
+If I would for example get 4 new emails, I would have to repeat this procedure 4 times.+
Yes, I'm not happy about it.
*I already did the following (using hints from other threads):*
Remove Mail.app, and /Library/Mail/
Installing Mail.app and the /Library/Mail/ folder again from the 10.5.0 DVD using Pacifist.
Updating the former installation with the 10.5.6 Combo update
Installing the Mail Update 10.5.6 afterwards
Removing the ~/Library/Preferences/com.apple.mail.plist
Restart
Launch Mail, setting up all accounts from scratch
Create a new user
Open Mail, setting up a new account
Reset Power Management
*So far, nothing did solve the problem.*
Next would be Archive and Install Mac OS X but I currently don't have the time for that procedure.
I have two Mac Pros (one at work, one at home) who both have this issue.
I believe, it's a bug though and will send this text to Apple via "Provide Mail Feedback" in Mail's Mail menu. -
Incoming SMTP issue with an ASA5512-X
I have reviewed numerous support discussions on this particular issue, but I am still unable to properly configure my ASA 5512-X to receive SMTP email. I can send email, and have access to all other services that I setup. I did create a network object for my Mail server and I am fairly certain this issue has something to do with my static NAT setup.
My current configuration is listed below...any assistance would be greatly appreciated.
I am new to the Cisco ASA appliance, and I am learning CLI as I go. I also have ADSM setup.
CONFIGURATION:
: Hardware: ASA5512, 4096 MB RAM, CPU Clarkdale 2792 MHz, 1 CPU (2 cores)
ASA Version 9.3(1)
names
interface GigabitEthernet0/0
nameif Verizon
security-level 0
ip address 100.39.18.94 255.255.255.0
interface GigabitEthernet0/1
description Local Norco Domain
nameif Norco.local
security-level 100
ip address 10.0.0.10 255.255.255.0
dhcprelay server 10.0.0.1
interface GigabitEthernet0/2
description SCE-DRAS SERVER
nameif SCE-DRAS
security-level 0
ip address 192.168.10.1 255.255.255.0
interface GigabitEthernet0/3
shutdown
no nameif
no security-level
no ip address
interface GigabitEthernet0/4
shutdown
no nameif
no security-level
no ip address
interface GigabitEthernet0/5
shutdown
no nameif
no security-level
no ip address
interface Management0/0
management-only
nameif management
security-level 100
ip address 192.168.1.1 255.255.255.0
boot system disk0:/asa931-smp-k8.bin
boot system disk0:/asa912-smp-k8.bin
ftp mode passive
dns domain-lookup Verizon
dns domain-lookup Norco.local
dns server-group DefaultDNS
name-server 10.0.0.1
name-server 68.238.96.12
name-server 68.238.64.12
domain-name norco.local
same-security-traffic permit inter-interface
same-security-traffic permit intra-interface
object service HTTP-19560
service tcp destination eq 19560
object service HTTP-65535
service tcp destination eq 65535
object service HTTP-8933
service tcp destination eq 8933
object service HTTP-8943
service tcp destination eq 8943
object service RTP
service udp destination range 19560 65535
object service SIP-TCP-8943
service tcp destination range 8933 8943
description IPPHONE - SIP
object service SIP-UDP-8943
service udp destination range 8933 8943
description IPPHONE - SIP
object service smtp
service tcp destination eq smtp
object network SMTP-SERVER
host 10.0.0.1
object-group network IPHONE-SERVERS
description VERIZON IP-PHONE SERVERS
network-object 128.177.14.0 255.255.255.0
network-object 128.177.36.0 255.255.255.0
network-object host 199.19.195.241
network-object host 199.19.195.243
network-object host 199.19.195.250
object-group service GENERAL-ACCESS tcp
description GENERAL SERVICES ACCESS
port-object eq ftp
port-object eq www
port-object eq https
port-object eq smtp
object-group service IP-PHONE-SERVICE
description PHONE SYSTEM ACCESS RULES
service-object object HTTP-19560
service-object object HTTP-65535
service-object object HTTP-8933
service-object object HTTP-8943
service-object object RTP
service-object object SIP-TCP-8943
service-object object SIP-UDP-8943
service-object tcp-udp destination eq 1025
service-object tcp destination eq www
service-object tcp destination eq https
service-object udp destination eq domain
service-object udp destination eq ntp
service-object tcp-udp destination eq domain
service-object tcp-udp destination eq www
service-object tcp-udp destination eq sip
service-object tcp destination eq domain
service-object tcp destination eq smtp
service-object tcp destination eq ssh
service-object tcp destination eq telnet
service-object udp destination eq dnsix
service-object udp destination eq www
object-group service General-TCP-UDP-Access
service-object tcp-udp destination eq domain
service-object tcp-udp destination eq www
service-object tcp destination eq domain
service-object tcp destination eq ftp
service-object tcp destination eq www
service-object tcp destination eq https
service-object udp destination eq www
service-object udp destination eq ntp
service-object udp destination eq radius
access-list Verizon_access_in extended permit object-group IP-PHONE-SERVICE object-group IPHONE-SERVERS any
access-list Verizon_access_in extended permit tcp any object SMTP-SERVER eq smtp
access-list Verizon_access_out extended permit object-group IP-PHONE-SERVICE any object-group IPHONE-SERVERS
access-list Verizon_access_out extended permit object-group General-TCP-UDP-Access any any
access-list Verizon_access_out extended permit tcp any any eq smtp
access-list SCE-DRAS_access_out extended permit ip any any
access-list SCE-DRAS_access_in extended permit ip any any
pager lines 24
logging enable
logging asdm informational
mtu Verizon 1500
mtu Norco.local 1500
mtu SCE-DRAS 1500
mtu management 1500
ip verify reverse-path interface Verizon
no failover
icmp unreachable rate-limit 1 burst-size 1
asdm image disk0:/asdm-731-101.bin
no asdm history enable
arp timeout 14400
no arp permit-nonconnected
nat (Norco.local,Verizon) source dynamic any interface
nat (SCE-DRAS,Verizon) source dynamic any interface
object network SMTP-SERVER
nat (Norco.local,Verizon) static interface service tcp smtp smtp
access-group Verizon_access_in in interface Verizon
access-group Verizon_access_out out interface Verizon
access-group SCE-DRAS_access_in in interface SCE-DRAS
access-group SCE-DRAS_access_out out interface SCE-DRAS
route Verizon 0.0.0.0 0.0.0.0 100.39.18.1 1
route Norco.local 10.10.0.0 255.255.255.0 10.0.0.7 2
timeout xlate 3:00:00
timeout pat-xlate 0:00:30
timeout conn 1:00:00 half-closed 0:10:00 udp 0:02:00 icmp 0:00:02
timeout sunrpc 0:10:00 h323 0:05:00 h225 1:00:00 mgcp 0:05:00 mgcp-pat 0:05:00
timeout sip 0:30:00 sip_media 0:02:00 sip-invite 0:03:00 sip-disconnect 0:02:00
timeout sip-provisional-media 0:02:00 uauth 0:05:00 absolute
timeout tcp-proxy-reassembly 0:01:00
timeout floating-conn 0:00:00
user-identity default-domain LOCAL
http server enable
http 192.168.1.0 255.255.255.0 management
http 10.0.0.0 255.255.255.0 Norco.local
no snmp-server location
no snmp-server contact
crypto ipsec security-association pmtu-aging infinite
crypto ca trustpoint _SmartCallHome_ServerCA
no validation-usage
crl configure
crypto ca trustpoint ASDM_Launcher_Access_TrustPoint_0
enrollment self
subject-name CN=10.0.0.10,CN=ciscoasa
crl configure
crypto ca trustpoint ASDM_Launcher_Access_TrustPoint_1
enrollment self
subject-name CN=10.0.0.10,CN=ciscoasa
crl configure
telnet timeout 5
ssh stricthostkeycheck
ssh timeout 5
ssh key-exchange group dh-group1-sha1
console timeout 0
dhcpd update dns both override
dhcpd address 192.168.10.2-192.168.10.5 SCE-DRAS
dhcpd dns 68.238.96.12 68.238.64.12 interface SCE-DRAS
dhcpd enable SCE-DRAS
dhcpd address 192.168.1.2-192.168.1.10 management
dhcpd enable management
dhcprelay information trust-all
threat-detection basic-threat
threat-detection scanning-threat
threat-detection statistics host
threat-detection statistics port
threat-detection statistics protocol
threat-detection statistics access-list
no threat-detection statistics tcp-intercept
ssl encryption rc4-sha1 aes128-sha1 aes256-sha1 3des-sha1
ssl trust-point ASDM_Launcher_Access_TrustPoint_1 Norco.local
ssl trust-point ASDM_Launcher_Access_TrustPoint_1 Norco.local vpnlb-ip
webvpn
anyconnect-essentials
no error-recovery disable
dynamic-access-policy-record DfltAccessPolicy
class-map inspection_default
match default-inspection-traffic
policy-map type inspect dns preset_dns_map
parameters
message-length maximum client auto
message-length maximum 512
policy-map global_policy
class inspection_default
inspect dns preset_dns_map
inspect ftp
inspect h323 h225
inspect h323 ras
inspect rsh
inspect rtsp
inspect sqlnet
inspect skinny
inspect sunrpc
inspect xdmcp
inspect sip
inspect netbios
inspect tftp
inspect ip-options
class class-default
user-statistics accounting
service-policy global_policy global
smtp-server 10.0.0.1
prompt hostname context
service call-homeHi Murali,
I ran the packet tracer routine...please let me know if I did this correctly...the results are below. The results indicate the traffic is denied by the implicit rule, but I'm not sure why....
ciscoasa(config)# packet-tracer input verizon tcp 209.85.213.176 smtp 100.39.18.94 smtp
Phase: 1
Type: ACCESS-LIST
Subtype:
Result: ALLOW
Config:
Implicit Rule
Additional Information:
MAC Access list
Phase: 2
Type: ROUTE-LOOKUP
Subtype: Resolve Egress Interface
Result: ALLOW
Config:
Additional Information:
in 100.39.18.94 255.255.255.255 identity
Phase: 3
Type: ROUTE-LOOKUP
Subtype: Resolve Egress Interface
Result: ALLOW
Config:
Additional Information:
in 0.0.0.0 0.0.0.0 via 100.39.18.1, Verizon
Phase: 4
Type: NAT
Subtype: per-session
Result: ALLOW
Config:
Additional Information:
Phase: 5
Type: ACCESS-LIST
Subtype:
Result: DROP
Config:
Implicit Rule
Additional Information:
Result:
input-interface: Verizon
input-status: up
input-line-status: up
output-interface: NP Identity Ifc
output-status: up
output-line-status: up
Action: drop
Drop-reason: (acl-drop) Flow is denied by configured rule -
I couldn't receive incoming call voice. Just could receive incoming call voice with ear speaker, headphone or built-in speaker.
Huh? What exactly is your issue?
-
How to remove caller name id for incoming call?
When i get an incoming call, along with the ringtone the device also says the name if the caller. How can i remove it and just have the ringtone and no name being called out.
klc757,
Thank you for reaching out us. We understand the importance of being able to set phone up as wanted. From you are describing it sounds like your phone is in driving mode. Do you see a steering wheel picture in the top bar of your phone? If so then pull the top bar down and close out of it. When did this issue begin? If you do not see these characteristics please fill us in on when this issue began and what you were doing when it happened.
Thank you,
TonyG_VZW
Follow us on Twitter @VZWSupport -
IOS 5.1 Update Issue: Contacts exist, but do not appear with incoming call or text
Alrighty; I just purchased the iPhone 4s and last night was prompted to upgrade to IOS 5.1. After doing so, I noticed today any incoming calls from one of my contacts would not display the name (the number was displayed as if it was an unknown call). However, text messages displayed contact information. I was playing around with it and based on suggestions elsewhere I added a '1' before some numbers. Upon doing this, an incoming call would display the name of the caller but incoming texts would not. A call to support directed me to call Verizon (my carrier) for 'some sort of three digit code' which would correct the issue; alas, Verizon had no clue. I'm at a loss and any help or suggestions would be greatly appreciated!
Is this forum only for the US? Maybe I'm intruding here by posting this?
Not only the Verizon version unfortunately. We have the same problem here in some parts of the Middle East. I have heard reports from Kuwait and Bahrain, and I'm located in Oman and the problem exists here as well. And no solution except downgrading the phone back to iOS 4.3.5. -
1.3.5.1 Updated, and Now Issue With Incoming Calls
Since updating to 1.3.5.1, any time I mute the ringer on an incoming call, it mutes the ringer until right before the missed call notification, but then plays the ringer again briefly. This defeats the purpose of being able to mute the ringer in the first place.
Post relates to: Pre p100eww (Sprint)This occurrs any time I press the volume rocker key to mute the ring of an incoming call. After the initial time when it played an extremely brief snippet of my ringtone (factory installed "Older Phone" ringtone), it has randomly switched between the short ringtone and a electronic squeal noise.
Since I know the slightest software change can kill a seemingly unrelated feature, I'll try to be as detailed as possible. For my phone setup, it is completely stock (phone no.5 after misc. slider/screen/charger door issues, haven't had time to change anything yet) My ringtone volume is set to maximum with vibrate also on, system sounds are also set to maximum with the default tone. I have no non-app catalog programs or patches installed. I use a Touchstone back cover, although the phone was not on the charger when I noticed the issue.
The only downloaded apps on the phone are Pandora, Tip 'em!, Stop Watch, Fandango, Good Food, Mileage Monitor and gDial Pro (which has not been set up yet). With the exception of Pandora, none of these apps have been used since I received this replacement phone. They were all just automatically downloaded because my previous phone had them. I have e-mail accounts through Cox, Yahoo, and Gmail, none of which are synced to contacts, calendars, messaging or tasks. I do have my Palm profile and EAS through my work in a full sync capacity.
Hope this helps.
Message Edited by kx250ryder on 01-07-2010 07:29 AM
Message Edited by kx250ryder on 01-07-2010 07:30 AM -
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...
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' ?
-
Cant reply to incoming call with text while on phone
Before IOS7 I could reply to an incoming call while on a call with a text " Cant talk right now I will call you later". With IOS7 that is not an option. I spoke with Apple tech support they said it was a Verizon issue. That Verizon has to update their system to be compatible with the update. I spoke with people who have iphone 5's with AT&T and Sprint and the have the same issue. This was a really nice feature and I wish it was someway to get it back. Can somebody help
I am not happy that they removed this feature either. Unfortunately, Apple seems to be changing many things for the worse since Steve Jobs is gone. User friendliness was a big priority to him, and this being one of those and also being efficient and possibly safer in the case that you may be on the phone and driving. You could, with one touch, let your second caller know that you were busy or you would call them back...this is not one of Apple's best decisions by far
-
Issue with transferring calls to VM for the correct DN Unity Connection 9.1.2
Hi all
I have been facing an issue with a Unity Connection Server v9.1.2. Every time an internal extension (assigned to VM profile and to Unity as a user) which is configured to be transferred to VM after 20 sec or so NoAN,is called , Unity treats the call as the extension of the calling party and not the called party. Furthermore , I dont know if this has any relation with the problem I am facing but when I check the voicemail port status in RTMT its seems like that regardless it is a direct call to the Unity from an extension (dial the pilot number or press the messages button on the IP phone) or a redirected call from an extension to Unity due to NoAN configuration, the Reason is Direct and the caller party number is always the extension initiated the call and not the one redirected to the Unity (second case).
I have changed the Use Last (Rather than First) Redirecting Number for Routing Incoming Call Unity parameter in the
System Settings > Advanced >Conversations -> checked and
Redirecting Diversion Header Delivery - Outbound CUCM parameter ->checked
in the SIP Trunk configuration used for the integraton of the CUCM with Unity but none of those seem to address this issue. Is there any guidelines you can give me to overcome this issue?
The servers I am using are
CUCM v9.1.2 BE
CUC v9.1.2 BE
Thank You in AdvanceHello again,
The Voice Mail Box Mask was blank before. However I tried XXXX (I use 4-digit extension for the VM pilot) but this did not seem to fix anything to the system....Same situation as before. Any more suggestions?
Thank you -
Hi...I have just obtained a secondhand iphone 5. I'm not technically minded and I don't understand 'geek speak' so please be gentle. I have done all I can think of to bottom this problem. I have switched on and off. Tried different o2 sims (including 2 brand new) that I know are fine, restored to factory settings, changed carrier from 'Auto' to O2 and spent ages on line with o2 chat but everything at their end is absolutely fine. Everything on the phone seems to be working apart from the phone aspect of things. All incoming calls go straight to voicemail and show on the phone as 'Cancelled calls' and when dialling out, I can key in the numbers and get the keypad tone (not dialling tone) but as soon as I press the green key to make the call, all that happens is the keypad pops back up onto the screen. The phone was bought on ebay as an o2 phone (the photo showed Giffgaff on the phone's screen but I understand this to be one and the same?) and as being 6 months old but I have put the serial number into the support website and...surprise surprise..,it must be at least twice that age as the warranty has expired. I have made 2 attempts to have a web chat with Apple Support, have reached the 'You're all set, we'll be with you in 2 mins' and have been left hanging there for over an hour on each occasion, so I've given that up as a bad job!! Any help will be much appreciated...and also, will Apple charge me £25 for each of the aborted chat attempts, please? Over to you...hopefully and many thanks in anticipation.
My phone works all fine when am in mumbai but as soon s i leave mumbai I am not able to make an otgoing call
This is entirely a carrier issue. If your carrier (airtel) doesn't provide service outside of Mumbia, this has nothing to do with your iPhone.
i feel i need to change my brand
This is a user-to-user tech support forum, not Apple. No one here cares at all about your threats.
NEVER post personal info in this public forum
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