Cisco 1760 and E1 to VoIP Gateway

Hello!
Recently we've developed a test VoIP network combined out of a verry differently branched devices. It's working quite well, excluding one part.
The problem occures when trying to call from a CCME Cisco IP phone registered in a gatekeeper to a MultiTechs Multivoip E1/VoIP gateway (which is connected to our PBX). In MvOIPs logs - I can focus Cisco Phone trying to connect using the correct codec, matching gateway prefix - BUT LEAVING NO IP ADDRESS OR H.323 ID. Mabey some ideas???
Network is H.323 based.
Thanks in advance.
Michael

Hi Micheal,
Can your PBX see the digits being passed to it?
Thanks,
Erik

Similar Messages

Cisco 1760 and VoIP gateway

Hello NetProf!
I'm repeating my request for help, mabey clearing a few things out.
1. We have a cisco 1760 router running CCME
2. Some connected to it cisco 79xx ip-phones
3. A third party gatekeeper - MikroTik router
4. Every device is logged in the gatekeeper, and every call i settled with the use of it.
5. We have a third party VoIP gateway - MultiTechs MultiVoIP - also registered in the gatekeeper.
6. Now the connections history:
a) when phoning from the gateway to cisco - everything works fine.
b) when phoning fromn cisco to the gateway - the connection failes - it looks like CISCO ISN'T FORWARDING ANY INDENTITY DATA - NO IP ADDRESS, AND H323-ID - GETS TO THE GATEWAY!!!!!
Please help!!
Michal Podoski

Check if there are some known issues documented here:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122relnt/1700/rn1700yj.htm

Cisco 1760-V Route Table & VoIP

I have connected a 1760-V to a T1 (CCS) which serves the connection to the PSTN, and configured the ethernet port which will facilitate communications to an IP PBX.
Assuming the IP address of the IP PBX and the IP address assigned to the router's ethernet interface are on the same subnet, and further assuuming that a dynamic protocol will not be configured/required on the router, what static route entry will be required, if any? I am assuming a default static route entry would have to be configured, suggesting that all IP traffic should be sent to the IP PBX.
Thanks,
Derek Fairley

I would tend to think this should work, if your 1760V is only dealing with PSTN traffic and not data traffic. With setting the default static route all traffic should be sent to your IP PBX as you mentioned.

Cisco Call Manager 5.1 and Cisco VoIP Gateways

Hi,
I have Cisco Call Manager 5.1 with Cisco 2800 series routers as a ISDN gateways and Cisco VG224/ATA 186 as POTS gateways. While Cisco Call Manager collects call statistics from 7900 series SCCP handsets managed by the Call Manager (number of VoIP packets, dropped frames etc.), it shows only one side of the call as it does not collect statistics from the Cisco routers (configured as H.323 gateways), the Cisco VG224 (with ports configured as H.323 gateways and as a SCCP devices) and Cisco ATA 186 (configured for SCCP).
Is the collection of these statistics possible? What needs to be done?
Thanks,
Paul

Update: Logging of QoS/Call Management Records (CMR) to Call Manager CDR database is not possible with H.323 gateways; only MGCP gateways:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_qanda_item09186a008020650a.shtml#qa4
Anyone know how to get the Cisco router gateway to log these statistics anyway using standard router logging mechanisms (e.g. syslog)?
Anyone know about calls to SCCP ports on Cisco VG224 or ATA 186?
Thanks,
Paul

Cisco BRAS 7206 VXR + voip gateways Micronet (part 2)

I'm sorry, I've closed previouse discussion - https://supportforums.cisco.com/message/3075308#3075308
The problem is actual yet.
Micronet voip gateway (SP5002) cant' up pppoe connection to CIsco 7206 vxr, after cisco uptime > 20 days.
I have attached wireshark dump.
IOS: c7200-advipservicesk9-mz.122-33.SRD4.bin
Config:
Current configuration : 12976 bytes
! Last configuration change at 16:14:26 KGST Tue May 18 2010 by tuxper
! NVRAM config last updated at 16:25:21 KGST Tue May 18 2010 by tuxper
version 12.2
service timestamps debug uptime
service timestamps log datetime msec localtime
service password-encryption
service counters max age 5
hostname Router_PPPoE
boot-start-marker
boot system disk2:c7200-advipservicesk9-mz.122-33.SRD4.bin
boot-end-marker
logging buffered 64000
no logging console
enable password 7 071D294A4D101F1F1E140C061C382D74716B
aaa new-model
aaa group server radius PPPOE-RADIUS-SERVER
server-private xxx.x.x.x auth-port 3812 acct-port 3813 key 7 yyyyyyyyy
aaa group server radius ISG-RADIUS-SERVER
server-private xx.x.x.x auth-port 4812 acct-port 4813 key 7 yyyyyyyyyy
aaa group server radius CISCO-AAA-SERVER
server-private xxx.x.x.x auth-port 2812 acct-port 2813 key 7 yyyyyyyyyyyyyy
aaa group server tacacs+ ACCT-TACACS-SERVER
server-private xx.x.x.x key 7 yyyyyyyyyyyyyyyyy
aaa authentication login default local group CISCO-AAA-SERVER
aaa authentication enable default group CISCO-AAA-SERVER enable
aaa authentication ppp PPPoE group PPPOE-RADIUS-SERVER
aaa authorization network PPPoE group PPPOE-RADIUS-SERVER
aaa authorization subscriber-service default local group ISG-RADIUS-SERVER
aaa accounting update periodic 1440
aaa accounting exec default start-stop group CISCO-AAA-SERVER
aaa accounting commands 1 default stop-only group ACCT-TACACS-SERVER
aaa accounting commands 15 default stop-only group ACCT-TACACS-SERVER
aaa accounting network PPPoE start-stop group PPPOE-RADIUS-SERVER
aaa accounting network ISG start-stop group ISG-RADIUS-SERVER
aaa server radius dynamic-author
client xxx.x.x.x server-key 7 yyyyyyyyyyyyy
auth-type all
ignore session-key
ignore server-key
aaa session-id common
clock timezone KGST 6
ip subnet-zero
ip source-route
ip cef
ip name-server xxxxxxxxx
ip name-server xxxxxxxxx
subscriber service password 7 xxxxxxxxxxxxxxxx
redirect server-group REDIR-SERVER
server ip xxx.xxx.xxx.xxx port 9999
multilink bundle-name authenticated
archive
log config
logging enable
notify syslog contenttype plaintext
   hidekeys
path ftp://[email protected]/bras7206-//[email protected]/bras7206-
write-memory
time-period 10080
username tuxper password 7 yyyyyyyyyyyy
username evrey password 7 yyyyyyyyyyyyy
bba-group pppoe global
virtual-template 1
sessions max limit 8000
sessions per-vlan limit 4000
interface Loopback0
ip address xx.xxx.xx.xx 255.255.255.255
interface GigabitEthernet0/1
description 7600_Uplink
ip address yy.yy.yy.yy 255.255.255.252
ip ospf network point-to-point
media-type rj45
speed 1000
duplex full
no negotiation auto
no cdp enable
interface GigabitEthernet0/2
description 7600_Uplink_pppoe_vlan
mtu 9000
no ip address
media-type rj45
speed auto
duplex auto
no negotiation auto
vlan-range dot1q 25 26
pppoe enable group global
   exit-vlan-config
vlan-range dot1q 300 399
pppoe enable group global
exit-vlan-config
vlan-range dot1q 401 461
   pppoe enable group global
exit-vlan-config
vlan-range dot1q 600 620
pppoe enable group global
exit-vlan-config
vlan-id dot1q 20
pppoe enable group global
exit-vlan-config
interface GigabitEthernet0/3
no ip address
shutdown
media-type rj45
speed auto
duplex auto
no negotiation auto
interface Virtual-Template1
description PPPoE
mtu 1492
ip unnumbered Loopback0
ppp authentication chap pap PPPoE
ppp authorization PPPoE
ppp accounting PPPoE
ppp eap refuse
ppp ms-chap refuse
ppp ms-chap-v2 refuse
ppp ipcp dns zzz.zzz.zzz.zzz zzz.zzz.zzz.zzz
router ospf 1
router-id yyy.yy.yy.yy
log-adjacency-changes
summary-address ii.iii.ii.0 255.255.254.0
summary-address ttt.ttt.ttt.0 255.255.255.0
summary-address ttt.ttt.ttt.0 255.255.255.0
summary-address ttt.ttt.ttt.0 255.255.255.0
summary-address ttt.ttt.ttt.0 255.255.254.0
redistribute connected subnets
redistribute static subnets
network uuu.uuu.uuu.uuu 0.0.0.3 area 0
default-information originate
ip classless
no ip http server
no ip http secure-server
!line con 0
stopbits 1
line aux 0
stopbits 1
line vty 0 4
session-timeout 30
exec-timeout 60 0
length 0
international
line vty 5 15
ntp clock-period 17179974
ntp peer vvv.vvv.vvv.vvv
ntp peer vvv.vvvv.vvv.vvv
end
P.S. I'm ready to provide any needed information.

I made config like this:
interface Virtual-Template1
description PPPoE
mtu 1492
ip unnumbered Loopback0
ppp authentication chap pap ms-chap ms-chap-v2 PPPoE
ppp authorization PPPoE
ppp accounting PPPoE
ppp eap refuse
ppp ipcp dns xxx.xx.xx.xx tt.tt.tt.tt
But the problem is actual.
I installed new IOS: c7200-advipservicesk9-mz.122-33.SRE1.bin
But 21 days is over, and I have the same problem.
I don't understand anything.

Setting Up VLAN and QoS for VOIP on SG200-18

We recently purchased the SG200-18 smart switch to replace a Netgear unmanaged switch. We're moving our phone service to VOIP through our local ISP as well.
I've currently got the VOIP phone plugged into Port 17 on the SG200-18 (it's a Grandstream cordless VOIP phone).
I want to put the VOIP phone on a separate VLAN from the rest of the network and optimize the QoS settings so that the VOIP phone has exceptional audio quality even during intense network traffic.
Here's my questions:
1. Do I need to adjust anything on the type of port for Port 17 (since it looks like some form of Combo port)?
2. How do I go about isolating the VOIP phone on it's own VLAN (I'm seeing VLAN and Voice VLAN settings, not sure which one to use; I tried setting a VLAN and broke Internet connectivity to the phone until I went in and removed it)?
3. Do I need to adjust any QoS settings on the switch to better optimize the VOIP phone?
A couple of additional questions about the GS200-18 in general:
1. Do I need to adjust any of the System Time Settings on the switch? I'm in Central Time.
2. Do I need to adjust any of the Green Ethernet/Energy Saving settings or should I stick with the defaults?
Also, a couple of "getting started" side questions to Cisco:
1. I've registered a My Cisco account. What do I need to do to register my switch with Cisco and associate it with my My Cisco account?
2. What are the benefits of taking out a Cisco Small Business Support Contract, and about how much would it cost on the SG200-18 (I ordered it from Provantage)? I'm curious to see if it's worth the money.
Here's my "specs":
Switch: SG200-18
VOIP phone: Grandstream DP715 and 710 expandable handsets
Plugged into: Port 17 on the SG200-18
ISP: Local ISP (Direclynx)
Connection type: 3M down/500k up DSL, moving to a wireless connection coming up which will give us faster speeds
VOIP backend provider: VOIP Innovations
Router: Apple Airport Extreme AC model (I run all Macs and iOS devices and OS X Server on the network, so using the Apple router makes setup easier, since it doesn't QoS, trying to QoS and VLAN at the switch level)
Thanks everyone!

Hello,
Lots of different questions here so I'll try to make sure I don't miss anything.
1. Do I need to adjust anything on the type of port for Port 17 (since it looks like some form of Combo port)?
The way the combo ports work is you can either use the SFP slot for a fiber connection or the copper ethernet port, but not both at the same time. Other then that they just function as normal network ports.
2. How do I go about isolating the VOIP phone on it's own VLAN (I'm seeing VLAN and Voice VLAN settings, not sure which one to use; I tried setting a VLAN and broke Internet connectivity to the phone until I went in and removed it)?
It sounds like you created the VLAN correctly and assigned the phone, however there wasn't anything doing any routing for that VLAN. You would need to have a VLAN capable router or a layer 3 switch so that something would act as the default gateway for the voice VLAN and route the traffic for you. Since there was nothing like this your phone lost it's connectivity to the internet when you placed it in the new VLAN. I don't think the Airport is VLAN capable, but we will come back to that.
3. Do I need to adjust any QoS settings on the switch to better optimize the VOIP phone?
Once you have a seperate VLAN setup for the phone properly you only have to tell the switch what your Auto Voice VLAN is going to be and it will automatically apply recommended QoS settings for the Voice VLAN and prioritize the voice traffic. There are ways to do this manually and even with the phone in the same VLAN however the are considerably more complicated.
1. Do I need to adjust any of the System Time Settings on the switch? I'm in Central Time.
The system time isn't always very important. You can set the correct time zone, however you should know the switch does not have a battery in it to keep track of time, so if/when it reboots or loses power the clock will reset. If you would like the switch to maintain accurate time you should setup an NTP server so the time is automatically updated from the internet. The switch will keep your timezone settings once you save them. Time is mostly important for logging and things like that, so you can configure it if you like but it is not necessary.
2. Do I need to adjust any of the Green Ethernet/Energy Saving settings or should I stick with the defaults?
Green ethernet simply reduces the power usage of the switch slightly, so unless you are having odd issues where ports are disconnecting, I would just leave them at the defaults.
1. I've registered a My Cisco account. What do I need to do to register my switch with Cisco and associate it with my My Cisco account?
There isn't really a way to associate your Small Business devices with your Cisco account. If you ever call in for technical support we will use your Cisco account and your serial number to create a support case, but even then they aren't linked together. If you decide to buy a support contract, that will be linked to your switch's S/N and your Cisco ID, so in a way that would associate them together. Devices being associated with Cisco accounts is something more common with Enterprise equipment, and mainly has to do with technical support cases.
2. What are the benefits of taking out a Cisco Small Business Support Contract, and about how much would it cost on the SG200-18 (I ordered it from Provantage)? I'm curious to see if it's worth the money.
There are a few advantages to a Support Contact. Your switch comes with a Limited Lifetime warranty that includes 1 year of technical support and return to factory hardware. With a service contract you get 3 years of technical support and next business day Advanced Replacement of the switch if it need to be replaced. I just did a quick google search, and it looks like a contract (part #CON-SBS-SVC2) costs about $50.
So there are a few other things to consider however.
As a frame of reference the average VOIP call uses about 64 - 128 kbps max.
Since you don't have a VLAN capable router or a layer 3 switch, a separate voice VLAN may not be an option. You also mention that the Apple Airport does not do QoS, meaning we will only be prioritizing the voice traffic while it is on the switch. When it is passed off to the Airport to be routed out to the internet all of the QoS settings will be lost, and normal network traffic will get the same priority as voice, since that is all up to the Airport.
With one phone the hassle of getting more equipment and setting up advanced QoS isn't really worth it, especially if the link to the internet isn't going to be participating in QoS.
One last thing I wanted to mention is you are switching to a wireless internet connection. I would ask them how their latency and jitter is, as these two network statistics greatly effect voice quality, and usually wireless performs worse when it comes to voice traffic.
I hope this information helps, if you have any more questions just let me know.
Thank you for choosing Cisco,
Christopher Ebert - Network Support Engineer
Cisco Small Business Support Center

Sip trunk between CUCM7.0 and third party VOIP provider

Hi all,
I'm looking for a solution/howto configuration for setting up a SIP trunk between CUCM7.0 and a SIP-VoIP provider.
Got SIP username, password and SIP-proxy IP from the provider.
I've done such a setup on CUCME a couple of times, but never on the CUCM.
Who can put me on right way?
Can it be done on the CUCM, or must an IOS-Device be used (got a PSTN-GW connected through H323 with CUCM)?
THanks for the hint,
Greets Norbert

Here we go.....
CONFIG (Version=7.1)
=====================
Version 7.1
Cisco Unified Communications Manager Express
! Calling nr. incoming
voice translation-rule 40
rule 1 /$.*$/ /0\1/
! Discard prefix (calling nr.)
voice translation-rule 190
rule 1 /^0$.*$/ /\1/
rule 2 /^9$.*$/ /\1/
! Mapping, internat to external nr.
voice translation-rule 191
rule 10 /^[1-9].*/ /xxxxEXTERNALxxxx/
! for call-forwarding
rule 15 /^0$.*$/ /\1/
! Mapping external to internal nr.
voice translation-rule 192
rule 2 /^xxxxxEXTERNALxxxx/ /4xx/
voice translation-profile TP_IN_SIP
translate calling 40
translate called 192
voice translation-profile TP_OUT_SIP
translate calling 191
translate called 190
dial-peer voice 2001 voip
corlist outgoing dialCORnoFax
description *** SIP-TRUNK (OUT) ***
translation-profile incoming TP_IN_SIP
translation-profile outgoing TP_OUT_SIP
max-conn 2
destination-pattern 9.T
session protocol sipv2
session target ipv4:2xx.xxx.xxx.xxx
session transport udp
! customer external nr. range (one dot at the and -> 0-9)
incoming called-number xxxxxxxx.
dtmf-relay rtp-nte
codec g711alaw
no vad
gateway
timer receive-rtp 1200
sip-ua
keepalive target ipv4:2xx.xxx.xxx.xxx
authentication username xxEXTERNAL NR.xxxxx password 7 111111111111111111111
calling-info pstn-to-sip from number set xxEXTERNAL NR.xxxxx
retry invite 2
retry response 2
retry bye 2
retry register 2
retry options 1
registrar ipv4:2xx.xxx.xxx.xxx expires 60
host-registrar
Greets,
Norbert
Hope this help......Please rate if helpful

Router 2901 as VoIP gateway

Hello,
I'm trying to configure a CISCO 2901 Integrated Services Router with a Module VWIC3-1MFT-T1/E1 as a VoIP gateway.
I'd like to know if is it possible, and if there is a especific document in CISCO documentation explaining it.
I have the following escenary:
Inbound calls:
PSTN (ISDN) --> CISCO 2901 --> Asterisk (SIP PBX)
Outbound calls
Asterisk(SIP PBX) --> CISCO 2901 --> PSTN (ISDN)
I have found some documents which explains how to configure a SIP Trunk with registration
http://www.cisco.com/en/US/docs/ios-xml/ios/voice/cube_sipsip/configuration/xe-3s/asr1000/voi-sip-trunk-reg.html
And how to configure the VWIC3 module
http://www.cisco.com/en/US/docs/routers/access/interfaces/software/feature/guide/vd-t1e1_vwic3.html
I'm very new with CISCO and I don't know how to route calls from the module to the Ethernet interface, and if I need to do anything else.
Thanks in advance!
David Pinedo
Presence Technology

For ISDN you need to configure your T1/E1, plenty of material and also references/examples on CSC, a quick search will provide some basic config
http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cucme/admin/configuration/guide/cmebasic.html
Search pri-group timeslots to get more examples.
Then you'll need to configure dial-peers for in/out calls, search cisco.com for dial-peer configuration and you'll get all the info you need on how to configure them.
HTH
java
if this helps, please rate
www.cisco.com/go/pdihelpdesk

Clear Channel Codec between Cisco GW and Patton Smartnode

Hi,
i have to establish an H.323 connection between an Cisco 3745 and an Patton
Smartnode 1200 with clear-channel codec to support ISDN-Data over VoIP.
PC with ISDN-Card---S0/PattonGW----H.323----3745/E1----PSTN
Software versions:
Cisco: SPServices 12.4(5a)
Patton SN1200: R3.20 2006-03-30 H323
The call establishment is sucessfull but i am not able pass any Data over
ISDN. Does anyone got this working?
Any help is welcome.
Regards,
Robert
Below are some Konfig sniplets:
Cisco:
controller E1 0/1/0
pri-group timeslots 1-31
interface Serial0/1/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn overlap-receiving
isdn incoming-voice voice
isdn send-alerting
isdn sending-complete
no cdp enable
dial-peer voice 10 pots
description ----- Towards PSTN -----
answer-address <removed>
destination-pattern 0T
progress_ind setup enable 3
fax rate disable
direct-inward-dial
port 0/1/0:15
dial-peer voice 473997246 voip
description ----- towards Patton GW -----
destination-pattern <removed>
progress_ind setup enable 3
modem passthrough nse codec g711alaw
session target ipv4:192.168.1.28
codec clear-channel bytes 80
ip qos dscp cs5 media
no vad
dial-peer voice 473997247 voip
description ----- Incoming Dial-Peer -----
modem passthrough nse codec g711alaw
incoming called-number <removed>
codec clear-channel bytes 80
ip qos dscp cs5 media
no vad
Patton:
profile voip VOIP
codec 1 transparent-cisco rx-length 20 tx-length 20
no high-pass-filter
no post-filter
no dtmf-relay
no dtmf-mute-encoder
context ip router
rtp-port-range 16384 32767
context cs switch
no digit-collection timeout
no address-completion timeout
interface h323 IF_H323
bind gateway GW_H323
route call dest-interface IF_S0_01
remote 192.168.254.30 1720
itc rx transparent
use profile voip VOIP
interface isdn IF_S0_01
route call dest-interface IF_H323
use profile tone-set Europe
no call-waiting
gateway h323 GW_H323
faststart
early-h245
h235-security version v2
no h235-security q931-auth-int
bind interface IF_IP_LAN router
no shutdown
use profile voip VOIP
port bri 0 1
clock auto
encapsulation q921
q921
protocol pmp
uni-side auto
encapsulation q931
q931
protocol dss1
uni-side net
encapsulation cc-isdn
bind interface IF_S0_01 switch

The SIP Media Inactivity Timer feature enables Cisco gateways to monitor and disconnect VoIP calls if no Real-Time Control Protocol (RTCP) packets are received within a configurable time period.
When RTCP reports are not received by a Cisco gateway, the SIP Media Inactivity Timer feature releases the hung session and its network resources in an orderly manner. These network resources include the gateway digital signal processor (DSP) and time-division multiplexing (TDM) channel resources that are utilized by the hung sessions. Because call signaling is sent to tear down the call, any stateful SIP proxies involved in the call are also notified to clear the state that they have associated with the hung session. The call is also cleared back through the TDM port so that any attached TDM switching equipment also clears its resources.
http://www.cisco.com/en/US/products/ps6441/prod_troubleshooting_guide09186a00801fa207.html#wp2077496

Cisco wireless and Apple Mac woes

Hello all,
I've been working with Cisco wireless and WLC's for a couple of years now but the recent onslaught of Apple Mac's is giving me heart burn. I've seen this at numerous sites now and need to throw it to eht community for guidance.
Basically we have had a number of instances where the Macs just fall off the wifi. Sometimes it's when they wake from sleep and other times when roaming between AP's (1131s with same SSID's). Our standard install is WPA2 and per ap local authentication. PC's work fine and never an issue.
We have completed a survey with a spectrum analyser and no RF interefence is present nor errors on the radio interface.
Questions:
- Is there a preferred Cisco config/setup for Mac's to work reliably? I've heard loads of rumors but nothing concrete and nor can I find anything specific.
- Should I be setting up WDS in case there is an authenticating issue.
- For those who are Mac gurus and happen to be reading. What Mac options we should look at?
This has all come to a head because the clients IT company who recommended the Macs (different from us doing the network infrastructure) are insisting that the problem is Cisco incompatibility and that we should rip out the Cisco kit and install airports (what tha!!!).
Thanks in advance for any pointers.
For those who like a config here it is .... Vanilla stuff really
Building configuration...
Current configuration : 2236 bytes
version 12.4
no service pad
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
hostname AP4
no logging console
enable secret xxxxxxxxxxxxxxxxx
no aaa new-model
dot11 syslog
dot11 ssid Home
   vlan 1
   authentication open
   authentication key-management wpa
   guest-mode
   mbssid guest-mode
   wpa-psk ascii xxxxxxxxxxxx
dot11 ssid avnet
   vlan 2
   authentication open
   authentication key-management wpa
   mbssid guest-mode
   wpa-psk ascii xxxxxxxxxxxxxxxx
username abcd password 1234
bridge irb
interface Dot11Radio0
no ip address
no ip route-cache
encryption vlan 1 mode ciphers tkip
encryption vlan 2 mode ciphers tkip
ssid Home
mbssid
speed basic-1.0 basic-2.0 5.5 11.0 6.0 9.0 12.0 18.0 24.0 36.0 48.0 54.0
channel 2412
station-role root
interface Dot11Radio0.1
encapsulation dot1Q 1 native
no ip route-cache
bridge-group 1
bridge-group 1 subscriber-loop-control
bridge-group 1 block-unknown-source
no bridge-group 1 source-learning
no bridge-group 1 unicast-flooding
bridge-group 1 spanning-disabled
interface Dot11Radio0.2
encapsulation dot1Q 2
no ip route-cache
bridge-group 2
bridge-group 2 subscriber-loop-control
bridge-group 2 block-unknown-source
no bridge-group 2 source-learning
no bridge-group 2 unicast-flooding
bridge-group 2 spanning-disabled
interface FastEthernet0
no ip address
no ip route-cache
duplex auto
speed auto
hold-queue 80 in
interface FastEthernet0.1
encapsulation dot1Q 1 native
no ip route-cache
bridge-group 1
no bridge-group 1 source-learning
bridge-group 1 spanning-disabled
interface FastEthernet0.2
encapsulation dot1Q 2
no ip route-cache
bridge-group 2
no bridge-group 2 source-learning
bridge-group 2 spanning-disabled
interface BVI1
ip address 192.168.10.54 255.255.255.0
no ip route-cache
ip default-gateway 192.168.10.1
no ip http server
no ip http secure-server
bridge 1 route ip
line con 0
line vty 0 4
login local
end

Yeah!! even i have come across multiple issue with MAC and Cisco.. these are the below settings which i normally do on the cisco gears and most of the times this solved the issue..
on the IOS AP disable Aironet Extentions and set the poer local and ofdm to max
no dot11 extension aironet
power local cck max
power local ofdm max
end
On the WLC, disable Aironet IE..
lemme know if this answered your question..
Regards
Surendra
====
Please dont forget to rate the posts which answered your question and mark it as answered or was helpfull

Urgent!!! Cisco ACE and asymetric routing assistance needed

I am wondering if someone can give me pointers on the cisco ACE
and asymetric routes. I've attached the diagram:
-Cisco IOS IP address is 192.168.15.4/24 and 4.1.1.4/24
-Firewall External interface is 192.168.15.1/24,
-Firewall Internal interface is 192.168.192.1/24,
-F5_BigIP External interface is 192.168.192.4/24,
-F5_BigIP Internal interface is 192.168.196.1/24 and 192.168.197.1/24,
-host_y has IP addresses of 192.168.196.10/24 and 192.168.197.10/24,
-Checkpoint has static route for 192.168.196.0/24 and 192.168.197.0/24
pointing to the F5_BigIP,
-host_y is dual-home to both VLAN_A and VLAN_B with the default
gateway on host_y pointing to VLAN_A which is 192.168.196.1,
-host_x CAN ssh/telnet/http/https to both of host_y IP addresses
of 192.168.196.10 and 192.168.197.10.
In other words, from host_x, when I try to connect to host_y
via IP address of 192.168.197.10, the traffics will go through VLAN_B
but the return traffics will go through VLAN_A. Everything
is working perfectly for me so far.
Now customer just replaces the F5_BigIP with Cisco ACE. Now,
I could not get it to work with Asymetric route with Cisco ACE. In
other words, from host_x, I can no longer ssh or telnet to host_y
via IP address of 192.168.197.10.
Anyone knows how to get asymetric route to work on Cisco ACE?
Thanks in advance.

That won't work because ACE uses the vlan id to distinguish between flows.
So when the response comes back on a different vlan, ACE can't find the flow it belongs to and it drops it.
Even if we could force it to accept the packet, ACE would then try to create a new flow for this packet and it will collide with the flow already existing on the frontend.
You would need to force your host to respond on the same vlan the traffic came in.
This could be done with client nat on ACE using different nat pool.
Gilles.

Can I add VOIP Gateway SPA2100-SU to the DMZ of Router BEFSR41 Ver 4 ?

I am having multiple problems with dropped calls; need to reboot my LinkSys VOIP Gateway and so on, and my VOIP provider has suggested the following: "Add your Gateway (the physical device I guess) to the DMZ of your router." We are trying to tell the Router to give this VOIP Gateway, which gets its IP address via DHCP from the router, to essentially give this Gateway carte blanche to any port or destination it wants. I have NO idea how to do this on the BEFSR41, which is a CA model with Version 4 appended to its model number ? Can anyone send me how to advice ? - Mike BRYAN Ottawa Canada PM me for my email.
(Edited post for guideline compliance. Thanks!)
Message Edited by JOHNDOE_06 on 01-22-2008 02:54 PM

First thing is to assign a static IP on the WAN / Internet side of your SPA-2100 so that you can set that IP to the DMZ of the BEFSR41. If you are not sure how to assign a static IP address on the spa2100 or how to open the web interface of the spa2100 and befsr41, then you better refer to the KB articles from www.linksys.com/kb. Just type in something like spa2100 or befsr41.
By the way, you may also enable NAT mapping and NAT Keep alive in your spa2100 aside from opening / forwarding ports in the befsr41 (check out Answer ID 5242). Lastly, enabling Send Resp To Src Port in the spa2100 is another option to try (this is under Voice > SIP > NAT Support Parameters).
It really requires much reading for us beginners to fully understand these devices. I am happy that I learned how to use the KBase site of Linksys since it gives me useful info about their products.

Setting up Cisco SLM248GT-NA switch for VoIP

Hello Everyone,
We have a Cisco slm248gt-na smart switch in our office that I am trying to configure to support and optimize for VoIP. We have desktops hooked up off the phones in most cases. Are there any config guidelines to optimize the switch so it has proper QOS set and VoIP gets higher priority over data.
Thanks, Kind Regards
Shabbir

Hello Tom,
Thanks for your reply.
We have a pretty straight forward setup. We have 15 Cisco SPA303 phones hooked up to switch and the desktop are connoted via the phone. We have a wireless LAN and a corporate server that servers as a file sharing/vpn type setup. Other than that there are no devices hooked up. We had to implement auto voice vlan feature but were still running into issues like occasional poor voice quality and one way audio type issues. However I did notice something in the configuration menu "Telephone OUI", could you please help us on how to implement that. We are trying to implement best practice to avoid voip related issues that we have been having.
We also see all the ports in trunk mode. Is that the recommended setting or should we change it to access mode?
Thanks, Kind Regards

Cisco 1760 with shdsl

I have a cisco 1760 router with one adsl wic and one shdsl wic (wic 1 shdsl v3).The router has detected the adsl wic but not the shdsl.I am using the following software:c1700-y7-mz.123-4.XG5.bin.I have tried several ios and used feature navigator to find ios but neither works.Here is output of sh version
Can somedy help ?

Already replied in WAN forum, please do not open duplicate threads.

Cisco 1760 router stays in rommon modus

I am having problems with a cisco 1760 router. When I turn the router on I am getting stuck in rommon mode. When I enter the command: "boot flash:c1700-sv8y7-mz.123-11.T3.bin" , the router will boot the ios image. But when i turn the router off and on, it will return to rommon mode. I have tried the following commands in rommon mode:
rommon 1 > confreg 0x2102
rommon 2 > reset
or
rommon 1 > confreg 0x142
rommon 2 > reset
without results. I get the following errors:
System Bootstrap, Version 12.2(7r)XM2, RELEASE SOFTWARE (fc1)
TAC Support: http://www.cisco.com/tac
Copyright (c) 2003 by cisco Systems, Inc.
Bad checksum on cookie structure, resorting to backup copy
Correcting primary cookie from backup
C1700 platform with 131072 Kbytes of main memory
loadprog: bad file magic number:      0x0
open: failed to find and/or load the bootloader: "flash:music-on-hold.au"
loadprog: error - on file open
boot: cannot load "c1700-sv8y7-mz.123-11.T3.bin"
System Bootstrap, Version 12.2(7r)XM2, RELEASE SOFTWARE (fc1)
TAC Support: http://www.cisco.com/tac
Copyright (c) 2003 by cisco Systems, Inc.
C1700 platform with 131072 Kbytes of main memory
loadprog: bad file magic number:      0x0
boot: cannot load "flash:"
System Bootstrap, Version 12.2(7r)XM2, RELEASE SOFTWARE (fc1)
TAC Support: http://www.cisco.com/tac
Copyright (c) 2003 by cisco Systems, Inc.
C1700 platform with 131072 Kbytes of main memory
loadprog: bad file magic number:      0x0
boot: cannot load "flash:"
System Bootstrap, Version 12.2(7r)XM2, RELEASE SOFTWARE (fc1)
TAC Support: http://www.cisco.com/tac
Copyright (c) 2003 by cisco Systems, Inc.
C1700 platform with 131072 Kbytes of main memory
rommon 1 >
rommon 1 >
Thanks,
Pieter

I would recommend posting in the netpro forums about this.
http://www.cisco.com/go/netpro
This site is for the Cisco Small Business Pro routers.
For your problem though, you might try looking at your boot system command in your running config to make sure that it is accurate.

Cisco 1760 and E1 to VoIP Gateway

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