VoIP issue with 881
Hi,
I have a strange issue with outgoing calles when connected to cisco 881.
I got an IPphone (cisco SPA502G) which is registered to our SipServer (Public IP Server). in the past we had the 871 unit and everything was OK.
since we replaced it with the new 881, I cannot make outgoing calles only incoming works.
the configuration is the same and very simple with no special NAT for the IP Phone:
interface FastEthernet4
description ### WAN ###
ip address X.X.X.X 255.255.255.248
ip nat outside
interface Vlan1
ip address 192.168.10.1 255.255.255.0
ip nat inside
ip route 0.0.0.0 0.0.0.0 X.X.X.X
ip nat inside source list 1 interface FastEthernet4 overload
access-list 1 remark LAN NAT
access-list 1 permit 192.168.10.0 0.0.0.255
Is there something specific which is needed for the VoIP on the new 800 serias ?
Thanx,
Arik
Hi,
I have a strange issue with outgoing calles when connected to cisco 881.
I got an IPphone (cisco SPA502G) which is registered to our SipServer (Public IP Server). in the past we had the 871 unit and everything was OK.
since we replaced it with the new 881, I cannot make outgoing calles only incoming works.
the configuration is the same and very simple with no special NAT for the IP Phone:
interface FastEthernet4
description ### WAN ###
ip address X.X.X.X 255.255.255.248
ip nat outside
interface Vlan1
ip address 192.168.10.1 255.255.255.0
ip nat inside
ip route 0.0.0.0 0.0.0.0 X.X.X.X
ip nat inside source list 1 interface FastEthernet4 overload
access-list 1 remark LAN NAT
access-list 1 permit 192.168.10.0 0.0.0.255
Is there something specific which is needed for the VoIP on the new 800 serias ?
Thanx,
Arik
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RV110W issues with VOIP service
Hi
I'm new to the forum and tearing my hair out here. We have an RV110W with a mix of computers and VOIP handsets.
Strange thing is that after 16 minutes of a VOIP call, the port seems to get swapped and the call hangs.
The VOIP provider, who is pretty worthless by the way - don't use Vivo Telecommunications - has decided that the problem resides at the firewall, They say:
The port swapping will be caused by the firewall, there are 1 of 2 causes: SIP Awareness; there are several things that can affect this most firewalls have SIP Alg but it can be disabled, also the NAT refresh time needs to be set correctly.
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The router has no settings for that. Make sure you have the latest firmware installed, there may have been an issue with the older firmware(s).
If the router is causing the issue then it has a bug. If you have another router, try it and see if you get the same result. If not, connect a VoIP handset directly to the ISP and make a call. If it breaks after 16 minutes, you know that the issue is with the ISP or VoIP provider, not the router. Another option is to disconnect everything from the RV110W (for security) except for a VoIP handset. Turn the firewall off on the router and try a call.
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www.cisco.com/go/sbsc
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Dell Wireless(DW1530, DW1540) issues with VoIP on 802.11n
We are having issues with VoIP, using Microsoft Lync, over wireless. We are experiencing static, jitter, dead air, and dropped words during calls. We have a Cisco wireless infrastructure consisting of a 5508 controller and 2602 access points. We're currently runng OS version 7.5.102.0, but the issues also occurred when running OS version 7.4.100.0.
On the client side, we've narrowed the issues down to Dell Laptops(Latitude E6430, Latitude D640, Latitude E6420) with Dell Wireless/Broadcom(DW1530, DW1540) network adapters that are 802.11n capable.
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All of the drivers are the latest offered from Dell.
If we disable high throughput (802.11n) on the access points and force the clients to connect with 802.11a or 802.11g, we don't experience any issues.
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I've seen some Intel chips that simply "misbehaves" when operating in 802.11n. Going to the laptop manufacturer's website to download the firmware is next to useless. Going to the NIC manufacturer's website to download the LATEST firmware is more reliable.
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Routing Issue with 3550 Switch
I am having an issue with routing with one of my Cisco 3550 switches. I know the 3550s are EoL but some of us have to work with what we have.
I am using a 3550 on either side of a Layer 2 link. The Layer 2 link is 2 Extreme Summit X-440 switches with Microwave between the switches. I have a VLAN configured on both switches and tagged on the ports connected to the Microwave. The 3550 switch on each end is configured for IP routing but I cannot pass traffic between the switches. If I unplug the switch on the local end and plug in a laptop, I can ping the switch on the remote end and access devices at the remote end.
I know this should work because I am doing the same thing over another Microwave link and Layer 2 link using another 3550 and a HP ProCurve at the remote end.
Here are the configs for each 3550:
Local end; Port Fa0/23 goes to the Remote Side. Port Fa0/24 goes to the rest of the network
Current configuration : 5417 bytes
! No configuration change since last restart
version 12.2
no service pad
service timestamps debug datetime localtime show-timezone
service timestamps log datetime localtime show-timezone
no service password-encryption
service sequence-numbers
hostname Brindley3550
enable secret 5 $1$3A.n$lzBUQg.fn4hJ7f0jEOqe71
no aaa new-model
clock timezone UTC -6
clock summer-time UTC recurring 1 Sun Apr 2:00 1 Sun Nov 2:00
mls qos map cos-dscp 0 8 16 26 32 46 48 56
mls qos min-reserve 5 170
mls qos min-reserve 6 10
mls qos min-reserve 7 65
mls qos min-reserve 8 26
mls qos
ip subnet-zero
ip routing
ip domain-name morgan911.net
ip name-server 1.2.150.11
ip name-server 1.2.150.5
spanning-tree mode pvst
no spanning-tree optimize bpdu transmission
spanning-tree extend system-id
vlan internal allocation policy ascending
interface FastEthernet0/1
switchport access vlan 18
switchport mode dynamic desirable
spanning-tree portfast
{Removed for Brevity}
|
interface FastEthernet0/7
switchport access vlan 13
switchport mode dynamic desirable
spanning-tree portfast
interface FastEthernet0/8
switchport access vlan 13
switchport mode dynamic desirable
spanning-tree portfast
{Removed for Brevity}
interface FastEthernet0/23
description To Gum Springs via Extreme P10
no switchport
ip address 1.2.147.1 255.255.255.252
speed 100
duplex full
interface FastEthernet0/24
description To Flint via Ceragon Eth 2
switchport trunk encapsulation dot1q
switchport mode trunk
speed 100
duplex full
mls qos trust cos
auto qos voip trust
wrr-queue bandwidth 20 1 80 1
wrr-queue min-reserve 1 5
wrr-queue min-reserve 2 6
wrr-queue min-reserve 3 7
wrr-queue min-reserve 4 8
wrr-queue cos-map 1 0 1 2 4
wrr-queue cos-map 3 3 6 7
wrr-queue cos-map 4 5
priority-queue out
spanning-tree link-type point-to-point
interface GigabitEthernet0/1
switchport trunk encapsulation dot1q
switchport mode trunk
interface GigabitEthernet0/2
switchport access vlan 10
switchport trunk native vlan 50
switchport mode dynamic desirable
spanning-tree portfast trunk
interface Vlan1
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ip default-gateway 1.2.145.1
ip classless
ip route 0.0.0.0 0.0.0.0 1.2.145.1
ip route 1.2.165.0 255.255.255.240 1.2.147.2
ip route 1.2.166.0 255.255.255.240 1.2.147.2
ip http server
snmp-server community public RO
snmp-server community public/RO RO
snmp-server location Brindlee Mountain Tower Site
snmp-server contact Jamey Wright
snmp-server enable traps snmp authentication linkdown linkup coldstart warmstart
snmp-server enable traps cluster
snmp-server enable traps entity
snmp-server enable traps envmon fan shutdown supply temperature
snmp-server enable traps vtp
snmp-server enable traps vlancreate
snmp-server enable traps vlandelete
snmp-server enable traps flash insertion removal
snmp-server enable traps port-security
snmp-server enable traps config
snmp-server enable traps syslog
snmp-server enable traps mac-notification
snmp-server enable traps vlan-membership
snmp-server host 1.2.150.100 public tty envmon syslog snmp
control-plane
ntp clock-period 17180143
ntp server 1.2.150.21
end
And this is the config for the remote end. Port Fa0/24 is the port for the link back to the local end.
Current configuration : 5058 bytes
version 12.2
no service pad
service timestamps debug datetime localtime show-timezone
service timestamps log datetime localtime show-timezone
no service password-encryption
service sequence-numbers
hostname GS3550
enable secret 5 $1$3A.n$lzBUQg.fn4hJ7f0jEOqe71
no aaa new-model
clock timezone UTC -6
clock summer-time UTC recurring
mls qos map cos-dscp 0 8 16 24 32 46 46 56
udld aggressive
ip subnet-zero
ip routing
ip domain-name morgan911.net
ip name-server 1.2.150.11
spanning-tree mode pvst
spanning-tree extend system-id
vlan internal allocation policy ascending
interface FastEthernet0/1
switchport access vlan 21
switchport mode dynamic desirable
spanning-tree portfast
interface FastEthernet0/2
switchport access vlan 21
switchport mode dynamic desirable
power inline delay shutdown 20 initial 300
spanning-tree portfast
{Removed for Brevity}
interface FastEthernet0/23
switchport access vlan 22
switchport trunk encapsulation dot1q
switchport mode trunk
speed 100
duplex full
spanning-tree portfast
interface FastEthernet0/24
description To Brindlee via Extreme P10
switchport mode dynamic desirable
(Is a member of VLAN 1)
speed 100
spanning-tree portfast
interface GigabitEthernet0/1
switchport trunk encapsulation dot1q
switchport mode trunk
interface GigabitEthernet0/2
switchport mode dynamic desirable
spanning-tree portfast
interface Vlan1
ip address 1.2.147.2 255.255.255.252
interface Vlan21
ip address 1.2.165.1 255.255.255.240
ip helper-address 1.2.150.11
ip helper-address 1.2.150.5
interface Vlan22
ip address 1.2.166.1 255.255.255.240
ip helper-address 1.2.150.5
ip helper-address 1.2.150.11
ip default-gateway 1.2.147.1
ip classless
ip route 0.0.0.0 0.0.0.0 1.2.147.1 10
ip http server
snmp-server community public RO
snmp-server enable traps snmp authentication linkdown linkup coldstart warmstart
snmp-server enable traps cluster
snmp-server enable traps entity
snmp-server enable traps envmon fan shutdown supply temperature
snmp-server enable traps vtp
snmp-server enable traps vlancreate
snmp-server enable traps vlandelete
snmp-server enable traps flash insertion removal
snmp-server enable traps port-security
snmp-server enable traps config
snmp-server enable traps hsrp
snmp-server enable traps bridge newroot topologychange
snmp-server enable traps syslog
snmp-server enable traps mac-notification
snmp-server enable traps vlan-membership
snmp-server host 1.2.150.100 public envmon syslog snmp
control-plane
ntp clock-period 17180192
ntp server 1.2.150.21 key 0 prefer
Ideas? Anything stand out as grossly wrong? I have worked on this for 2 days and am at a loss.
Thanks
JameySorry for the delay in replying. Other items at the office took priority over this project. I tried that and no change. I pulled the switch from the remote site and took it back to the local end and connected the switches with a crossover cable and everything works fine. I have pretty much determined that it is an issue with the config in one of the Extreme switches. The config in those look pretty normal but there are a few things I am unsure of. Guess I'll see if there is a similar site for Extreme gear.
Thanks
Jamey -
Issue with LPCOR on CME 10.5
Dear All,
I am facing issues with LPCOR configuration on CME 10.5. For International calls the Authentication Prompts triggers some times and some times doen not.
Also when a local call is dialed the Authentication Prompt is triggered some times.Below is the config and debug logs. Need your help to resolve this.
voice lpcor enable
voice lpcor custom
group 10 endusers
group 11 pstn
voice lpcor policy endusers
service fac
accept endusers fac
accept pstn fac
voice lpcor policy pstn
service fac
accept endusers fac
accept pstn fac
application
package auth
param passwd-prompt flash:enter_pin.au
param max-retries 0
param abort-digit *
param term-digit #
param user-prompt flash:enter_account.au
param passwd 12345
param max-digits 32
interface GigabitEthernet0/1.1
encapsulation dot1Q 1 native
ip address 10.25.76.1 255.255.255.0
interface GigabitEthernet0/1.201
encapsulation dot1Q 201
ip address 10.25.77.1 255.255.255.0
voice-port 0/0/0
lpcor outgoing pstn
trunk-group ALL_FXO 1
supervisory disconnect dualtone mid-call
supervisory custom-cptone 2n-gsm
no battery-reversal
input gain -6
output attenuation -3
cptone SA
timeouts call-disconnect 1
timeouts wait-release 1
timing sup-disconnect 50
connection plar 5040
caller-id enable
cable-detect
dial-peer cor custom
name local
name longdistance
name 911
name Internal
name fac-int
name user-fac
dial-peer cor list local
member local
dial-peer cor list call-local
member local
dial-peer cor list call-longdistance
member longdistance
dial-peer cor list user1
member local
member 911
dial-peer cor list user2
member local
member longdistance
member 911
member user-fac
dial-peer cor list user3
member 911
dial-peer cor list call-911
member 911
dial-peer cor list call-internal
member Internal
dial-peer cor list fac-int
member local
member 911
member fac-int
dial-peer cor list user-fac
member user-fac
dial-peer voice 96 pots
trunkgroup ALL_FXO
corlist outgoing call-911
destination-pattern 9[2-6]......
forward-digits 7
dial-peer voice 901 pots
trunkgroup ALL_FXO
corlist outgoing call-911
destination-pattern 901[2-4,6-8].......
forward-digits 10
dial-peer voice 800 pots
trunkgroup ALL_FXO
destination-pattern 9800T
prefix 800
dial-peer voice 900 pots
destination-pattern 9T
port 0/0/3
prefix 9
dial-peer voice 11 pots
destination-pattern 901........
port 0/0/3
forward-digits 10
dial-peer voice 9051 pots
trunkgroup ALL_FXO
corlist outgoing call-local
destination-pattern 905........
forward-digits 10
dial-peer voice 19 pots
trunkgroup ALL_FXO
corlist outgoing fac-int
destination-pattern 900T
translate-outgoing called 1
forward-digits all
dial-peer voice 20 voip
description International calling
service clid_authen_collect
destination-pattern 900T
lpcor outgoing pstn
session target ipv4:10.25.76.1
incoming called-number 9T
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
ephone-dn 1
number 4121
name John
corlist incoming fac-int
ephone 1
lpcor type local
lpcor incoming endusers
mac-address E0D1.730A.21DE
ephone-template 2
type 7942
button 1:1
voice register dn 33
number 4163
call-forward b2bua busy 5000
call-forward b2bua noan 5000 timeout 20
call-forward b2bua unregistered 5000
allow watch
name Joseph
mwi
voice register pool 33
busy-trigger-per-button 4
id mac BC67.1C31.C8AA
type 7821
number 1 dn 33
cor incoming fac-int 1 4163
dtmf-relay rtp-nte
codec g711ulaw
transfer max-length 4
Debug Logs
DAMAC-CME-ANOUD#DEBUg VOIce lpcor all
voip lpcor all debugging is on
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#term
DAMAC-CME-ANOUD#terminal i
DAMAC-CME-ANOUD#terminal i
Apr 12 16:22:39.825: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1, ConnectionId F692C420E06611E4BB0CE7FDC5486EA5, SetupTime 16:22:35.615 UTC Sun Apr 12 2015, PeerAddress 4130, PeerSubAddress , DisconnectCause 10 , DisconnectText normal call clearing (16), ConnectTime 16:22:39.825 UTC Sun Apr 12 2015, DisconnectTime 16:22:39.825 UTC Sun Apr 12 2015, CallOrigin 2, ChargedUnits 0, InfoType 2, TransmitPackets 0, TransmitBytes 0, ReceivePackets 0, ReceiveBytes 0
Apr 12 16:22:39.825: %VOIPAAA-5-VOIP_FEAT_HISTORY: FEAT_VSA=fn:TWC,ft:04/12/2015 16:22:35.609,cgn:4130,cdn:,frs:0,fid:2599,fcid:F692C420E06611E4BB0CE7FDC5486EA5,legID:284C,bguid:F692C420E06611E4BB0CE7FDC5486EA5mon
DAMAC-CME-ANOUD#terminal imon
^
% Invalid input detected at '^' marker.
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
Apr 12 16:22:44.089: //-1/xxxxxxxxxxxx/LPCOR/lpcor_get_index_by_name:
lpcor endusers
Apr 12 16:22:44.089: //-1/xxxxxxxxxxxx/LPCOR/lpcor_get_index_by_name:
lpcor endusers index 10
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#show debug
VOIP LPCOR:
debug voip lpcor error call is ON (filter is OFF)
debug voip lpcor error call informational is ON (filter is OFF)
debug voip lpcor error software is ON
debug voip lpcor error software informational is ON
debug voip lpcor detail is ON (filter is OFF)
debug voip lpcor function is ON (filter is OFF)
debug voip lpcor inout is ON (filter is OFF)
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
Apr 12 16:23:22.889: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1, ConnectionId FBA1532AE06611E4BB10E7FDC5486EA5, SetupTime 16:22:44.089 UTC Sun Apr 12 2015, PeerAddress 4130, PeerSubAddress , DisconnectCause 10 , DisconnectText normal call clearing (16), ConnectTime 16:23:02.009 UTC Sun Apr 12 2015, DisconnectTime 16:23:22.889 UTC Sun Apr 12 2015, CallOrigin 2, ChargedUnits 0, InfoType 2, TransmitPackets 0, TransmitBytes 0, ReceivePackets 1038, ReceiveBytes 166080
Apr 12 16:23:22.889: %VOIPAAA-5-VOIP_FEAT_HISTORY: FEAT_VSA=fn:TWC,ft:04/12/2015 16:22:44.093,cgn:4130,cdn:,frs:0,fid:2600,fcid:FBA1532AE06611E4BB10E7FDC5486EA5,legID:284D,bguid:FBA1532AE06611E4BB10E7FDC5486EA5
Apr 12 16:23:22.905: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1, ConnectionId FBA1532AE06611E4BB10E7FDC5486EA5, SetupTime 16:22:57.795 UTC Sun Apr 12 2015, PeerAddress 0097150107659, PeerSubAddress , DisconnectCause 10 , DisconnectText normal call clearing (16), ConnectTime 16:23:02.015 UTC Sun Apr 12 2015, DisconnectTime 16:23:22.905 UTC Sun Apr 12 2015, CallOrigin 1, ChargedUnits 0, InfoType 2, TransmitPackets 1038, TransmitBytes 174384, ReceivePackets 1043, ReceiveBytes 166880
Apr 12 16:23:22.905: %VOIPAAA-5-VOIP_FEAT_HISTORY: FEAT_VSA=fn:TWC,ft:04/12/2015 16:22:57.785,cgn:4130,cdn:0097150107659,frs:0,fid:2601,fcid:FBA1532AE06611E4BB10E7FDC5486EA5,legID:284E,bguid:FBA1532AE06611E4BB10E7FDC5486EA5
Apr 12 16:23:25.317: //-1/xxxxxxxxxxxx/LPCOR/lpcor_get_index_by_name:
lpcor endusers
Apr 12 16:23:25.317: //-1/xxxxxxxxxxxx/LPCOR/lpcor_get_index_by_name:
lpcor endusers index 10
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#un all
All possible debugging has been turned off
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#!ok just send me these logs
DAMAC-CME-ANOUD#!i have to move from here
Apr 12 16:24:02.153: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1, ConnectionId 14343755E06711E4BB16E7FDC5486EA5, SetupTime 16:23:25.323 UTC Sun Apr 12 2015, PeerAddress 4130, PeerSubAddress , DisconnectCause 10 , DisconnectText normal call clearing (16), ConnectTime 16:23:43.393 UTC Sun Apr 12 2015, DisconnectTime 16:24:02.153 UTC Sun Apr 12 2015, CallOrigin 2, ChargedUnits 0, InfoType 2, TransmitPackets 0, TransmitBytes 0, ReceivePackets 930, ReceiveBytes 148800
Apr 12 16:24:02.153: %VOIPAAA-5-VOIP_FEAT_HISTORY: FEAT_VSA=fn:Tnow
DAMAC-CME-ANOUD#\WC,ft:04/12/2015 16:23:25.321,cgn:4130,cdn:,frs:0,fid:2602,fcid:14343755E06711E4BB16E7FDC5486EA5,legID:2850,bguid:14343755E06711E4BB16E7FDC5486EA5
Apr 12 16:24:02.169: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1, ConnectionId 14343755E06711E4BB16E7FDC5486EA5, SetupTime 16:23:39.169 UTC Sun Apr 12 2015, PeerAddress 0097150107659, PeerSubAddress , DisconnectCause 10 , DisconnectText normal call clearing (16), ConnectTime 16:23:43.389 UTC Sun Apr 12 2015, DisconnectTime 16:24:02.169 UTC Sun Apr 12 2015, CallOrigin 1, ChargedUnits 0, InfoType 2, TransmitPackets 930, TransmitBytes 156240, ReceivePackets 937, ReceiveBytes 149920
Apr 12 16:24:02.169: %VOIPAAA-5-VOIP_FEAT_HISTORY: FEAT_VSA=fn:TWC,ft:04/12/2015 16:23:39.169,cgn:4130,cdn:0097150107659,frs:0,fid:2603,fcid:14343755E06711E4BB16E7FDC5486EA5,legID:2851,bguid:14343755E06711E4BB16E7FDC5486EA5We have come across this issue today in 10.9.5 (so affects 10.9.4 as well) but it was occurring in Sydney as well with a client and for me in Melbourne.
-
Issue with instant ringback when using sip trunk to SP
Hi all,
We use CUCM 8.0.2.
We have a SIP trunk to a SP connected via one of our Cisco 2911 routers configured as a CUBE.
Cisco IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version 15.0(1)M3, RELEASE SOFTWARE (fc2)
c2900-universalk9-mz.SPA.150-1.M3.bin
Cisco CISCO2911/K9 (revision 1.0)
Technology Package License Information for Module:'c2900'
Technology Technology-package
Current Type
ipbase ipbasek9 Permanent
security securityk9 Permanent
uc uck9 Permanent
data None None
We also have several ISDN lines that run out via various Cisco routers configured as H323 gateways.
We use 7945 and CIPC for our phones.
We're having an issue with calls going via the SIP trunk where we hear ringing instantly after dialling - but before the actual device at the other end starts ringing (considerable difference).
Using the SIP trunk: If I make a call to my mobile phone - I hear ringing instantly - about 3 rings before my mobile phone actually starts ringing - undesireable.
Using H323 gateway: If I make a call to my mobile phone - I hear silence for a bit - then ringing when the mobile starts ringing - desired.
Using SIP trunk: If I make a call to a landline that is ready - it rings instantly for at least 1 ring - before the actual phone I'm calling starts ringing - undesireable.
Using H323 gateway: There is a momentary pause before hearing ringing on my phone and the phone I dialled - desired.
Using SIP trunk: If I make a call to a landline that is off-hook (with no call-waiting/etc.) - it rings once and then returns the busy signal (the worst issue) - undesireable.
Using H323 gateway: There is a momentary pause before hearing busy signal - desired.
Phone to phone internally (same network): Operates as expected (instantly rings locally and on the phone I'm calling). Between phones that utilise the SIP trunk and phones that utilise the H323 gateways within the same network - communication is instant and as expected.
Any ideas why this happens and how to stop it?
I want it to not ring until the situation is known and that it can provide the appropriate feedback (ringing/busy/etc.).
Some possibly relevant config (note that there is a known bug with this IOS that meant I had to declare the codec in each dial-peer as the voice class would not work):
voice service voip
address-hiding
mode border-element
allow-connections sip to sip
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
header-passing error-passthru
early-offer forced
midcall-signaling passthru
interface GigabitEthernet0/0
ip address x.x.x.x 255.255.255.252
ip access-group acl.SIP-IN in
no ip redirects
no ip unreachables
ip verify unicast reverse-path
ip virtual-reassembly
duplex full
speed 100
no cdp enable
gateway
timer receive-rtp 1200
sip-ua
connection-reuse
gatekeeper
shutdown
dial-peer voice 1 voip
description *** INBOUND CALLS FROM CARRIER ***
translation-profile incoming SIPTRUNK-INCOMING
session protocol sipv2
incoming called-number #blah blah#
dtmf-relay rtp-nte
codec g711alaw
ip qos dscp cs5 media
no vad
dial-peer voice 61 voip
description **** WA, SA AND NT NUMBERS ****
destination-pattern 0[8]........
session protocol sipv2
session target ipv4:<MY SP's SIP SERVER>
incoming called-number 0[8]........
dtmf-relay rtp-nte
codec g711alaw
ip qos dscp cs5 media
no vad
dial-peer voice 81 voip
description **** MOBILE NUMBERS ****
destination-pattern 0[4]........
session protocol sipv2
session target ipv4:<MY SP's SIP SERVER>
incoming called-number 0[4]........
dtmf-relay rtp-nte
codec g711alaw
ip qos dscp cs5 media
no vad
dial-peer voice 500 voip
description *** INBOUND SIP TRUNK TO CUCM PUB ***
translation-profile outgoing SIPTRUNK-CALLING-ADD-0
preference 1
destination-pattern 5[12]..
session protocol sipv2
session target ipv4:<OUR CUCM PUBLISHER IP>
dtmf-relay rtp-nte
codec g711alaw
ip qos dscp cs5 media
no vad
Any help or a point in the right direction would be greatly appreciated.
Cheers,
BrettI ended up resolving this issue as follows:
In CUCM, under Device > Device Settings > SIP Profile.
I modifed the profile relevant to my SIP trunk, under the "Trunk Specific Configuration", I set "SIP Rel1XX Options" from "Disabled" to "Send PRACK if 1xx Contains SDP".
Now, I get the expected delay before hearing ringback.
Solved! -
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# -
Satellite L500 and issues with wireless connection
Hello people
Ill just throw my problem out there.
First of all my details and info which may help you help me!
Satellite L500 running windows 7 (note: same issues when Vista was installed)
Latest wireless drivers as well as latest drivers for everything else.
My wireless router is a billion voip enabled router 7000 series its quite a good one for when I got it about 2 yrs ago and works fine.
Basically the problem is the wireless on my Satellite L500 keeps dropping out. Im no more than 20m away at most, right now Im 10m away with 1 wall in the way. I tried to get to a website and it says connection dropped out or unable to find host.
However the other laptops in the same room same spot at the same time has no issues at all. Never has.
As soon as I go into the same room as my router it works again. Sometimes works if I just stand on the opposite wall.
So basically Im thinking my wireless card in my laptop is either faulty or just no good?
Can u replace the wireless card with a different one? Maybe the Toshiba ones are just no good? Ive read so many problems with these wireless cards in Toshibas...
Anyone have any ideas? Have I missed any info u need? Ill repost again in a day or so with the types of wireless devices in each laptop.Hi
At the moment I doubt that your WLAN card is faulty. Before you exchange it you should also update your BIOS and not only the WLAN card driver. Both updates you should download from the Toshiba website:
http://eu.computers.toshiba-europe.com > Support & Downloads > Download Drivers
I had a similar issue with my Toshiba notebook and WLAN connection. The problem was the WLAN power saving mode in Windows Power Management. There you can also choose a power saving mode for the WLAN card. Disable it and choose Maximum Performance. -
About the voip issue being not available(not a sol...
Hi Friends
I have formatted my phone.Reinstalled the firmware and installed sip_voip3.0 settings v1 (yes v1) With v2 it was bouncing me of the menu but this time the menu was available.I did the settings and i saw that my phone was connected.There was a globe sign at standby screen.Everything seemed perfect until i tried my first call.The phone says "net calls enabled" but they are not..I can not call anywhere and also i cant be called.It does not work.Maybe someone may continue and find a solution from here...Just wanted to share with you all suffering from voip issue.
Thanksyes this worked for me - I was able to place a SIP Call using 'SIP_VoIP_3_0_Settings_v1_01_en.sis'
Available at:
http://www.forum.nokia.com/info/sw.nokia.com/id/105455c9-654b-427f-99c7-202aac27a8d8/SIP_VoIP_3_0_Se...
The funny thing i was going to try this earlier - but I was too lazy to sign up an account. The only thing that sucks is there is no way to start or stop the service with the 'internet tel' so once it is running it is running...
Well guess this proves Nokia was lying - should have done this LONG ago.
One note if you have a + at the start of your contacts phone number you might have issues.
I love my purple screen, 1/2 day battery, sketchy touch screen and $800 price tag - I call it my Lumia 900 -
I recently purchased a Linksys WRT55AG (v2, I'm sure a V1 with custom firmware would do the trick) as an upgrade from my WRT54G (which was the greatest router I have ever had with the Alchemy firmware!). My only issue with the 55AG is that I cannot get any VOIP device to function behind it. I have tried 3 devices from 2 different providers with no luck (Vonage - Motorola VT2442, Cisco ATA 186; Sunrocket - Gizmo). Placing the device in DMZ doesn’t work, port forwarding is useless. I've contacted both the VOIP providers and Linksys with no resolution. I am running the latest firmware, and it doesn’t look as if Linksys is going to be providing anymore firmware updates for this model. Anyone run into a similar problem? Any help would be much appreciated!
Thanks,
EI am now having the same problem with the same router and a lightyear voip box. if you figured it out, please let me know.
Kurstin -
IChat issue with Siemens SE567 Router
I have just signed up with a new ISP and they insist I have to use the Siemens SE567 router. It works fine for the internet most of the time, but I can not do screen sharing or video chat. The strangest thing is I could do both with my last ISP the only difference now is the router and ISP. I have tried all different configurations and been on tech support with both Apple and my ISP to not avail. Anyone else having any issues with this router?
Thanks,
PamHi,
I am not sure if this was you I was Texting to yesterday (20th).
I linked the person to the Port Forward site
Their Set up pages were blue and white as opposed to this grey and Orange.
They were set to UPnP and NAPT as it is called on this device had no ports set.
The Computer was using Tiger with the Firewall Off.
Stealth (Ping Blocking) was also Off
We tried calling from both ends.
The result was an Error 7 at the Siemens end until we made sure the Quicktime Setting was done.
A second computer that had no camera had iChat turned Off to prevent any confusion about IP addresses.
The User did not think the ISP Offered VoIP and had no Vonage device (an issue with ichat 2 and 3)
My Error 8 log did not identify that the far end was using Tiger and iChat 3
(Should list port 5060 as SIP port)
Firewall in the Siemens was set to Low.
DMZ was Off.
There seem to be no Ping Blocking (Not that the log got that far) or DoS or SPI settings.
File Sharing in System Preferences > Sharing is not an issue in Tiger so not checked.
From reviewing the Chat I can see I did not check if System Preferences > Quicktime > Streaming was at 1.5Mbps
If you have to change this and iChat is Open restart iChat.
Also in the Firewall on the Mac in the Advanced Button is Block UDP Connection Active ?
IF so it will need deselecting as most of iChat's A/V Stuff is on UDP.
The Port Forward set of Pic suggest that the one they had was Branded for Telus as an ISP so it could be that this has slightly different software/firmware.
I suggested posting here as I had run out of iDeas on this device.
8:09 PM Thursday; January 21, 2010
Please, if posting Logs, do not post any Log info after the line "Binary Images for iChat" -
Is it safe to disable SPI Firewall Protection on your home network router due to a VOIP issue?
I've been having connection issues with my VOIP phone, which is connected through my router.
I have a home office and I use a VOIP phone for work. I’ve been having constant issues with calls getting disconnected in the middle of a conversation. I have contacted the phone service to troubleshoot and support said it was because of a Firewall setting on my Smart WiFi EA6900 router.
He disabled my SPI Firewall settings. I am concerned with this because this is for my home network and not just for my work phone. I am not very experienced with advanced configuration of a router so I need help. Is this change in my router settings safe for my other systems on the network?
I contacted Linksys support and they said my router has "built in protection" but when I pressed for additional information on what that mean he said "security will less secured compared when the SPI firewall is disabled." and again, when I pressed for a better explanation he said "he firewall primarily protects the router, and secondarily protects your computer."
This does not tell me whether the disabling of SPI is a safe option or not. Especially after I'm seeing posts (basic Google search) about how you should not disable SPI firewall.
I also read this article:
http://community.linksys.com/t5/Wireless-Routers/Voip-provider-says-turn-off-SPI-Firewall-is-that-sa...Hi. Disabling any security features of the router has a disadvantage in any situation. But doing so, resolves most challenges experienced using a VOIP device behind a Linksys router. What router are we discussing here?
I have checked other resources which gave information to forward these port numbers:
(*) port 5000-5500 for UDP and TCP
(*) port 10000-20000 for UDP
Also, ensure that the router's firmware is already the latest. It would be much better to work together with your VOIP provider on this. -
Hi,
I have a v2 HH, which for some reason, will seemingly drop the ability to perform VoIP calls after a few hours.
This is a problem as the VoIP number is the main phone number we use in the home.
Oddly, the old v1 HH I have, doesn't have this problem, but I'm reluctant to use this as I require setting a device on my network to DMZ - which the v1 doesn't support unless you're willing to forsake VoIP capability.
Going back to the v2 - it would appear that after a time, picking up the handset, will produce no dial tone, and we're unable to recieve any incoming calls.
Anyone have any further info on this? is it a known issue with the v2's? (which, if it is, would beg the question of why wasn't there a product recall)
Failing that, is it at all possible to use a seperate ADSL router - say, a netgear model, and have the v1 HH connected to that solely as a VoIP gateway?Hi Psybernoid,
Thank you for posting. Having read you post it sounds like to me that you might need to reconfigure your Hub. If you take a look at the link below it has some information on how to do so.
http://bt.custhelp.com/cgi-bin/bt.cfg/php/enduser/cci/bt_adp.php?p_sid=OjlKH9Wj&cat_lvl1=345&cat_lvl...
Please post back an let me know if that has helped.
Cheers,
Paddy
BTCare Community Mod
If we have asked you to email us with your details, please make sure you are logged in to the forum, otherwise you will not be able to see our ‘Contact Us’ link within our profiles.
We are sorry but we are unable to deal with service/account queries via the private message(PM) function so please don't PM your account info, we need to deal with this via our email account :-) -
Issues with Creative Cloud for teams deployment workflow
The Adobe Creative Cloud for teams IT Deployment Guide lists out steps for IT admins to deploy the CS6 applications and then have their end-users license the trial software with their Adobe IDs once they have been invited to the team. There are two major issues with this document.
First, the media that is on the FTP is not for North American English. We are working to get that posted on the FTP site ASAP. In the meantime, you can find the CS6 MC media from: http://www.adobe.com/downloads/
[Note: Getting media from that page requires the use of the Adobe Download Assistant which is very consumer focused. Sorry about that.]
Second, in order to have the ability to login properly with a Creative Cloud for Teams account the system needs to have the latest copy of Adobe Application Manager installed. If you do not do this step the end user will be prompted for a serial number.
Unfortunately the Adobe Application Manager can’t be packaged with AAMEE nor is it a native installer. I know, I know! Here are the links to the Adobe Application Manager installers:
Windows: http://www.adobe.com/support/downloads/detail.jsp?ftpID=4773
Mac: http://www.adobe.com/support/downloads/detail.jsp?ftpID=4774
It can be installed from command line by:
Win: <Path to Setup.exe>Set-up.exe –mode=silent –action=install
Mac: <path to ASU> /ASU/Install.app/Contents/MacOS/Install –mode=silent –action=install
Jody Rodgers | Sr. Product Manager | Creative Cloud for Enterprise | Adobe SystemsHi Boncker,
I see that you have an active Subscription under your account . Please launch any of the installed product and when you get the trial prompt , please click on License this software and then Enter the Adobe Id & Password for the account that you have accepted the invite .
Please do let us know if that worked for you or not .
Cheers,
Kartikay Sharma -
Issue with magsafe/charging (blinks green, amber, off)
Having a quirky issue with my mid-2009 Macbook Pro.
A month or so ago, I began having an issue with my macbook charging. When the computer is up and running and I plug in the magsafe it will say "Calculating..." and then say "Not Charging" and then switch to battery use. It constantly does this as long as the magsafe is connected. The lights on the magsafe will constantly cycle from green to amber to no light.
At first, it would do it a few times then it would start charging. Now it constantly does this without ever stopping. I was editing video the other day for a while and it never stopped the cycle.
Thinking it was the battery, I took it into the Apple Store last week and got a new battery because it had been saying "Service battery" for a while and the battery would only last about 20 minutes on a charge. So I just thought the battery finally refused to charge.
But the problem still exists.
The magsafe is new (only 3 months old) and here's the quirky thing: the computer will charge if it's asleep (lid closed) or shut down completely. No blinking lights at all. It will charge as long as it's asleep or shut down. So I don't think it's the charger. But as soon as you hit the power button to starting booting up, it will blink.
I have reset the SMC a couple times actually, but no change.
Is there something software/firmware related that I haven't tried? Any insight or suggestions would be greatly appreciated before I take it back to the Apple Store (which is about an hour away).
Thanks!Try resetting the system management controller
http://support.apple.com/kb/HT3964?viewlocale=en_US
If that did nothing for you, try resetting the NVRAM
https://support.apple.com/kb/HT1379
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