Cisco 2610xm NM2V FXO FXS

I have 2 cisco 2610xm gw one with fxs and another one with fxo i want to originate call from the fxs and terminating with the fxo connected to pstn line. But the IOS does not support Voice-port command. What IOS is recommended for voip on the 2610xm and where can i get a sample configuration for terminating calls to pstn with the fxo. Thnaks

Hi there any ios ip plus code will do, better if u can go for some latest version on 12.3(GD) train .but do remember that your box have enough DRAM and Flash space to acommodate the new ios code.
and as far as the config samples do a search using keywords like configuring fxo or fxs cards in cisco.com site that will yield lots of sample config templates ,which will be helpful to configure as per ur requirement.
regds

Similar Messages

FXO & FXS Card

Hello Guys ,
I have installed VIC-2FXS and VIC-2FXO Card on my Cisco 2651, but my router is not detecting the Voice cards.
I have seen the output of Show Version and Show Diag.
If i am installing any other WIC-2T it is detecting.
Can you Please let me know what might be the problem.
Waiting for your reply,
Regards,
MAX

First, let's examine the current system:
1-you have an 2651 XM
2-using an ios that is voice capable(c2600-ipvoice-mz.123-13)
3-the platform supports one network module and two interface cards
Secondly, let's examine what you did:
1-you installed a nm-4a/s (that means four synch/asynch serial wan interface) into the NM slot of 2651XM.
2-you installed a fxo and a fxs voice interface card into two built in interface card slots.
Lastly, see what is the problem:
1-nm-4a/s works without any problem
2-system does not recognise fxo or fxs cards
So let's see what is the reason:
1-2651xm does not support fxo or fxs interface cards into the built in interface card slots.(where you try to install fxo&fxs!!!)
2-2651xm supports voice interface cards by using a voice network module(instead you use nm-4a/s to get asynch/synch wan interfaces!!!)
At the end, let's say what must be the right hardware configuration:
1-one nm-hd-2v(to install into nm slot of 2651xm)(also this nm provide enough dsp for vics)
2-one vic2-2fxo and one vic2-2fxs(this cards must be installed into nm-hd-2v)
3-two wic-2a/s (this cards will be installed into built in interface card slots of 2651xm)
So you need:
one nm-hd-2v + one vic2-2fxo + one vic2-2fxs + two wic-2a/s; on the other hand, nm-4a/s will be unnecessary
Thanks

Radius Alive Packet Cisco 2610XM

Hello.
I have a Cisco 2610XM with firmware version c2600-ipvoicek9-mz.124-15.T5.bin. Could someone tell me how to suppress the ALIVE accounting packets coming to my Radius server?.
I have been trying using the command "no aaa accounting update" but it keeps sending the ALIVE accounting packets.
Thank in advance,
Regards,
Ricardo.-

This is the link to the forum, just in case:
http://forums.cisco.com/eforum/servlet/NetProf?page=netprof&forum=Security&topic=AAA&CommCmd=MB%3Fcmd%3Ddisplay_messages%26mode%3Dnew%26location%3D.ee6e1fe
Btw can you try this and see if it works:
no gw-accounting aaa
Regards
Farrukh

- FXO- FXS- call transfer not working...

We have cisco2811 with c2800nm-adventerprisek9-mz.124-16 and 1 FXO + 2 FXS ports.
at this moment i use CME B-ACD to auto answering and forward call to internal fxs port.
problem - this call, terminated to fxs, cann't be transferred by using flash button on analog phone(to transfer call from fxs to fxs port i use app-h450-transfer.2.0.0.9.tcl)
Can somebody tall me WHAT to use to make transfer feature working??
my config in attachment

Configuring Feature Mode for SCCP Analog Phones
This task configures feature mode for SCCP analog phones on the voice gateway and enables SCCP analog phones to invoke features using feature access codes after hook flash.
Prerequisites
You must first configure full consultation transfer on Cisco CallManager using the transfer-system full-consult command in order for feature mode to work on the voice gateway.
SUMMARY STEPS
1. enable
2. configure terminal
3. stcapp call-control mode feature
my IOS says that didn't know "stcapp" command....

US 500 Series FXO/FXS

Hello all. I just need to clarify/ask a question. From my discussions with Cisco tech support, the maximum number of FXO and FXS ports I can have on a UC 500 series is 8 of each, using a VIC card for 4 more FXO and a SPA8800 for 4 more FXS.
Does anyone know of anyway to increase this? I have a client who needs to run with 12 FXO and 12 FXS to make a functioning system.
Does anyone have any other solutions I can deliver? (And please do not go into BRI or SIP; it's not going to happen.)
Thanks all.

Hello Kris,
If you are using UC560-FXO and add two additional FXO VICs - VIC2-4FXO you will have a total of 12 FXO ports.
Then add SPA8000 will give you additional 8 FXS ports - total 12.
This is how you add SPA8000 to the UC.
https://supportforums.cisco.com/docs/DOC-9465
For FXS also you may use ATA186.
HTH,
Alex
*Please rate helpful posts

Tieing 2 key systems together with fxo/fxs and 1760 routers

Hello,
I need some programming help from someone good on voice. I've got two offices that I'm trying to tie the phone systems together with 2x Cisco 1760 routers each with 2x PVDM-256K-4 1 DSP Modules. I've got the layout below and am basically looking to do two things.
First, I would like ext. 210 from the first site to dial a co group â1â or directly access a âCO lineâ that is connected to the Cisco and get dial tone to be able to dial the directory number for a âCO lineâ with the same setup at the second site and have it able to be answered like a normal call and be transferred.
The second connection I would like is to have ext. 210 be able to dial locally to one of site 1's analog single line extensions and have the Ciscos make a connection through to site 2 and go off-hook on one of the analog single line extensions of site 2 in order to get a site 2 dial tone and be able to dial locally @ site 2 to any extension, or dial one of site 2's co groups or directory number for one of site 2's real CO lines and place a âlocalâ call to the outside world from site 2's lines.
Obviously this process would all be reversed for site 2 accessing site 1. I've come across a couple of documents, like ID: 15405, and a section of a VoIP Configuration guide labeled OL-1070-01 and have some command structure available, but the concept of how it all takes place and should be configured is a little fuzzy.
Thank you,
Mark

OK, let me simplify things. I think I'm putting way too much thought into it all. I've got site A and site B. Site A (currently for testing) has a single line extension from Site A's key system plugged into port 0 in fxo card in slot 2. Site A will have a patch from port 0 in fxs card in slot 3 to a CO line on the key system. Site B has the same setup. Both have fa 0/0 configured with IP addresses on the same network (just to simulate the connections - later I will actually move these to two separate internet feeds for more advanced testing).
Currently for testing I have disconnected the fxs patches to the phone systems and just have a regular analog phone plugged in. When my phone plugged into Site A goes off hook, I get dial-tone from the extension hooked up to Site B (which is the exact way I want it). When my analog phone is plugged into Site B (port 0 of fxs card in slot 3) and goes off hook, it will ring port 0 of fxs card in slot 3 of Site A. This I don't understand. If I can get both to behave like Site A, I'd be happy.
I need to know if this makes sense to anyone on how I want this to operate? Is it achievable?
Here's my base config on it (Site A first, then Site B):
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Simmering
boot-start-marker
boot-end-marker
enable secret 5 $1$F/AM$ige2qFh9lVD6uNubE.qm80
no aaa new-model
voice-card 2
voice-card 3
ip cef
interface FastEthernet0/0
ip address 192.168.254.30 255.255.255.0
speed auto
no ip http server
no ip http secure-server
control-plane
voice-port 2/0
connection plar opx 290
voice-port 2/1
connection plar opx 291
voice-port 2/2
voice-port 2/3
voice-port 3/0
connection plar 190
voice-port 3/1
connection plar 191
voice-port 3/2
voice-port 3/3
dial-peer voice 280 pots
destination-pattern 280
port 2/0
dial-peer voice 281 pots
destination-pattern 281
port 2/1
dial-peer voice 290 voip
destination-pattern 29
session target ipv4:192.168.254.40
dial-peer voice 180 pots
destination-pattern 180
port 3/0
dial-peer voice 181 pots
destination-pattern 181
port 3/1
dial-peer voice 190 voip
destination-pattern 19
session target ipv4:192.168.254.40
line con 0
logging synchronous
line aux 0
line vty 0 4
password Corazon64789
logging synchronous
login
transport input telnet
end
SITE B:
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Sigma
boot-start-marker
boot-end-marker
enable secret 5 $1$1H58$6.VsVieyi4srcC7v4Lndv0
no aaa new-model
voice-card 2
voice-card 3
ip cef
interface FastEthernet0/0
ip address 192.168.254.40 255.255.255.0
speed auto
no ip http server
no ip http secure-server
control-plane
voice-port 2/0
connection plar opx 280
voice-port 2/1
connection plar opx 281
voice-port 2/2
voice-port 2/3
voice-port 3/0
connection plar 180
voice-port 3/1
connection plar 181
voice-port 3/2
voice-port 3/3
dial-peer voice 290 pots
destination-pattern 290
port 3/0
dial-peer voice 291 pots
destination-pattern 291
port 3/1
dial-peer voice 280 voip
destination-pattern 28
session target ipv4:192.168.254.30
dial-peer voice 190 pots
destination-pattern 190
port 2/0
dial-peer voice 191 pots
destination-pattern 191
port 2/1
dial-peer voice 180 voip
destination-pattern 18
session target ipv4:192.168.254.30
line con 0
logging synchronous
line aux 0
line vty 0 4
password Corazon64789
logging synchronous
login
transport input telnet
end

FXO - FXS PSTN emulation for testing lab

Hi netpros,
For testing purposes. I am wondering whether it is possible to simulate PSTN by connecting routers FX0<->FXS ports as per the diagram. I will be purchasing E1 cards in several weeks but for now I am trying to get PSTN emulation using what I have got. Also ..do I need to have FXO ports on the 'PSTN emulator' device as well or having FXS connected to the edges router's FXO will be fine? Do I need to use PLAR to redirect the incoming calls from PSTN to the destination analog phone ?
Your help is much appreciated !!!
NOTE: I have tried getting some help on the Training forum without any luck

Hi,
Yes It will work.But you have to give connection plar under FXO ports.
HTH.
Please Rate if it helps

FXO/FXS IOS COMMAND ISSUES

Hello,
I've got 2 1760 routers with each a VIC2-4FX0 and VIC-FXS/DID cards. I've got 128mb of system RAM in them, running IOS c1700-advsecurityk9-mz.124-3.bin, which is supposed to be a voice ios with advanced feat. pack according to Cisco's website. I've also got 2x of the needed dsp's (can't remember model number, but its only 1 dsp chip on it - again cisco said 1 dsp would handle 4 channels, unless we're using the high-end codecs. My issue is that I cannot issue any of the voice-port commands. I don't think the hardware is somehow installed correctly. When I issue a "show hardware", they are not listed, however, when I capture a boot, they are listed as "card in slot 2/ card in slot 3" where the router allocates memory to devices.
Can anyone please help? And by the way, these are default no config, all I have done on the one is give a hostname, fastethernet ip and some basic password sec. and that's it.
HERE'S THE BOOT:
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
program load complete, entry point: 0x80008000, size: 0xcbd968 Self decompressing the image : #################################################
# [OK]
Smart Init is enabled
smart init is sizing iomem
ID MEMORY_REQ TYPE
MainBoard 0X00027A80 1760
0X000F3BB0 public buffer pools
0X00211000 public particle pools
0X0003B100 DSP Buffers
0X004C 0X00000000 Card in slot 2
0X003A 0X00000000 Card in slot 3
TOTAL: 0X00367730
If any of the above Memory Requirements are "UNKNOWN", you may be using an unsupported configuration or there is a software problem and system operation may be compromised.
AND HERE'S SHOW HARDWARE:
Simmering#sh hardware
Cisco IOS Software, C1700 Software (C1700-ADVSECURITYK9-M), Version 12.4(3), REL EASE SOFTWARE (fc2) Technical Support: http://www.cisco.com/techsupport Copyright (c) 1986-2005 by Cisco Systems, Inc.
Compiled Fri 22-Jul-05 12:04 by hqluong
ROM: System Bootstrap, Version 12.2(7r)XM2, RELEASE SOFTWARE (fc1)
Simmering uptime is 4 days, 16 hours, 17 minutes System returned to ROM by reload System image file is "flash:c1700-advsecurityk9-mz.124-3.bin"
This product contains cryptographic features and is subject to United States and local country laws governing import, export, transfer and use. Delivery of Cisco cryptographic products does not imply third-party authority to import, export, distribute or use encryption.
Importers, exporters, distributors and users are responsible for compliance with U.S. and local country laws. By using this product you agree to comply with applicable laws and regulations. If you are unable to comply with U.S. and local laws, return this product immediately.
A summary of U.S. laws governing Cisco cryptographic products may be found at:
http://www.cisco.com/wwl/export/crypto/tool/stqrg.html
If you require further assistance please contact us by sending email to [email protected].
Cisco 1760 (MPC860P) processor (revision 0x500) with 114688K/16384K bytes of mem ory.
Processor board ID FOC08252LAR (617609820), with hardware revision 0000 MPC860P processor: part number 5, mask 2
1 FastEthernet interface
32K bytes of NVRAM.
32768K bytes of processor board System flash (Read/Write)
Configuration register is 0x2102
Simmering#
Thank you,
Mark

I'm not that of an in-depth cisco programmer yet, I'm still working on my ccna - 1/2 way there, but this is what show diag gave me:
Simmering#show diag
Slot 0:
C1760 1FE VE 4SLOT DV Mainboard Port adapter, 1 port
Port adapter is analyzed
Port adapter insertion time unknown
EEPROM contents at hardware discovery:
Hardware Revision : 5.0
PCB Serial Number : FOC08252LAR
Part Number : 73-7167-05
Board Revision : B0
Fab Version : 04
Product (FRU) Number : CISCO1760
EEPROM format version 4
EEPROM contents (hex):
0x00: 04 FF 40 03 16 41 05 00 C1 8B 46 4F 43 30 38 32
0x10: 35 32 4C 41 52 82 49 1B FF 05 42 42 30 02 04 FF
0x20: FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF
0x30: FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF
0x40: FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF
0x50: FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF
0x60: FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF
0x70: FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF
Simmering#
I don't understand that because there is nothing in slot 0, and there are cards in slots 2 and 3. According to the cisco ios selector or whatever their cross-reference is said this ios has the voice/voip features, but if you have another ios version you think I should be using, please by all means let me know.
Thanks,
Mark

Cisco gk + cisco gw + many korea fxs gw

All experts
1)will cisco 3620 ios gatekeeper can support how many concurrent h323 voip call? 60?
2) the davolink (korea) fxs gw make call to cisco 2620 (T1) isdn router, but the call can only hold 20sec, then it dropped suddenly. (other taiwan fxs gw wont happen like this)
debug isdn q931
Cause i = 0x80E6 - Recovery on timer expiry
i had tried isdn T310= 10sec, 60 sec, 100sec also not ok....what is the problem??

Feb 11 19:42:32.211: H225 NONSTD OUTGOING ENCODE BUFFER::= 60 01040001 1E301E02 82881E02 82811C26 9E810003 67746400 00001B43 50472C0D 0A50524E 2C697364 6E2A2C2C 4E543130 302C0D0A 0D0A
Feb 11 19:42:32.215:
Feb 11 19:42:32.215: H225.0 OUTGOING PDU ::=
value H323_UserInformation ::=
h323-uu-pdu
h323-message-body alerting :
protocolIdentifier { 0 0 8 2250 0 4 }
destinationInfo
vendor
vendor
t35CountryCode 181
t35Extension 0
manufacturerCode 18
gateway
protocol
voice :
}, h323 :
mc FALSE
undefinedNode FALSE
callIdentifier
guid '0287762C4E26255A5634343434EF0000'H
multipleCalls FALSE
maintainConnection FALSE
h245Tunneling FALSE
nonStandardControl
nonStandardIdentifier h221NonStandard :
t35CountryCode 181
t35Extension 0
manufacturerCode 18
data '60010400011E301E0282881E0282811C269E8100...'H
tunnelledSignallingMessage
tunnelledProtocolID
id tunnelledProtocolAlternateID :
protocolType "gtd"
messageContent
'4350472C0D0A50524E2C6973646E2A2C2C4E5431...'H
tunnellingRequired NULL
Feb 11 19:42:32.239: H225.0 OUTGOING ENCODE BUFFER::= 23 80060008 914A0004 2800B500 00124002 38501B0C 00110002 87762C4E 26255A56 34343434 EF000001 00010010 A801003E 0140B500 00123760 01040001 1E301E02 82881E02 82811C26 9E810003 67746400 00001B43 50472C0D 0A50524E 2C697364 6E2A2C2C 4E543130 302C0D0A 0D0A2242 04677464 011B4350 472C0D0A 50524E2C 6973646E 2A2C2C4E 54313030 2C0D0A0D 0A
Feb 11 19:42:32.251:
Feb 11 19:42:33.671: ISDN Se1/1:23 Q931: RX <- CONNECT pd = 8 callref = 0x8646
Feb 11 19:42:33.687: ISDN Se1/1:23 Q931: TX -> CONNECT_ACK pd = 8 callref = 0x0646
Feb 11 19:42:33.695: H225 NONSTD OUTGOING PDU ::=
value H323_UU_NonStdInfo ::=
version 4
protoParam qsigNonStdInfo :
iei 28
rawMesg '1C269E8100036774640000001B414E4D2C0D0A50...'H
Feb 11 19:42:33.699: H225 NONSTD OUTGOING ENCODE BUFFER::= 60 01040001 1C281C26 9E810003 67746400 00001B41 4E4D2C0D 0A50524E 2C697364 6E2A2C2C 4E543130 302C0D0A 0D0A
Feb 11 19:42:33.699:
Feb 11 19:42:33.703: H225.0 OUTGOING PDU ::=
value H323_UserInformation ::=
h323-uu-pdu
h323-message-body connect :
protocolIdentifier { 0 0 8 2250 0 4 }
destinationInfo
vendor
vendor
t35CountryCode 181
t35Extension 0
manufacturerCode 18
gateway
protocol
voice :
}, h323 :
mc FALSE
undefinedNode FALSE
conferenceID '0287762C4E2A62645634343434EF0000'H
callIdentifier
guid '0287762C4E26255A5634343434EF0000'H
multipleCalls FALSE
maintainConnection FALSE
h245Tunneling FALSE
nonStandardControl
nonStandardIdentifier h221NonStandard :
t35CountryCode 181
t35Extension 0
manufacturerCode 18
data '60010400011C281C269E8100036774640000001B...'H
tunnelledSignallingMessage
tunnelledProtocolID
id tunnelledProtocolAlternateID :
protocolType "gtd"
messageContent
'414E4D2C0D0A50524E2C6973646E2A2C2C4E5431...'H
tunnellingRequired NULL
Feb 11 19:42:33.731: H225.0 OUTGOING ENCODE BUFFER::= 22 80060008 914A0004 2800B500 00124002 38500287 762C4E2A 62645634 343434EF 00001D0C 00110002 87762C4E 26255A56 34343434 EF000001 00010010 A8010036 0140B500 00122F60 01040001 1C281C26 9E810003 67746400 00001B41 4E4D2C0D 0A50524E 2C697364 6E2A2C2C 4E543130 302C0D0A 0D0A2242 04677464 011B414E 4D2C0D0A 50524E2C 6973646E 2A2C2C4E 54313030 2C0D0A0D 0A
Feb 11 19:42:33.747:
Feb 11 19:42:33.751: H225 NONSTD OUTGOING PDU ::=
value IRRperCallnonStandardInfo ::=
startTime 1076499753
Feb 11 19:42:33.751: H225 NONSTD OUTGOING ENCODE BUFFER::= 70 402A1529
Feb 11 19:42:33.755:
Feb 11 19:42:33.755: RAS OUTGOING PDU ::=
value RasMessage ::= infoRequestResponse :
requestSeqNum 11474
endpointType
vendor
vendor
t35CountryCode 181
t35Extension 0
manufacturerCode 18
gateway
protocol
voice :
}, h323 :
mc FALSE
undefinedNode FALSE
endpointIdentifier {"82691D700000006C"}
rasAddress ipAddress :
ip 'CA40F944'H
port 53559
callSignalAddress
ipAddress :
ip 'CA40F944'H
port 1720
endpointAlias
h323-ID : {"85444"}
perCallInfo
nonStandardData
nonStandardIdentifier h221NonStandard :
t35CountryCode 181
t35Extension 0
manufacturerCode 18
data '70402A1529'H
callReferenceValue 3806
conferenceID '0287762C4E2A62645634343434EF0000'H
originator FALSE
h245
callSignaling
callType pointToPoint : NULL
bandWidth 160
callModel direct : NULL
callIdentifier
guid '0287762C4E26255A5634343434EF0000'H
needResponse FALSE
unsolicited TRUE
Feb 11 19:42:33.779: RAS OUTGOING ENCODE BUFFER::= 5A C02CD128 00B50000 12400238 501E0038 00320036 00390031 00440037 00300030 00300030 00300030 00300036 004300CA 40F944D1 370100CA 40F94406 B8014004 00380035 00340034 003401E1 00B50000 12057040 2A15290E DE028776 2C4E2A62 64563434 3434EF00 000000A0 03C00011 00028776 2C4E2625 5A563434 3434EF00 000E2401 000180
Feb 11 19:42:33.795:
Feb 11 19:42:33.799: H225.0 OUTGOING PDU ::=
value H323_UserInformation ::=
h323-uu-pdu
h323-message-body facility :
protocolIdentifier { 0 0 8 2250 0 4 }
conferenceID '0287762C4E2A62645634343434EF0000'H
reason startH245 : NULL
callIdentifier
guid '0287762C4E26255A5634343434EF0000'H
h245Address ipAddress :
ip 'CA40F944'H
port 14174
multipleCalls FALSE
maintainConnection FALSE
h245Tunneling FALSE
Feb 11 19:42:33.807: H225.0 OUTGOING ENCODE BUFFER::= 26 90060008 914A0004 0287762C 4E2A6264 56343434 34EF0000 8101001F 05801100 0287762C 4E26255A 56343434 34EF0000 0700CA40 F944375E 01000100 10800100
Feb 11 19:42:33.815:
Feb 11 19:42:38.171: RAS OUTGOING PDU ::=
value RasMessage ::= registrationRequest :
requestSeqNum 11475
protocolIdentifier { 0 0 8 2250 0 4 }
discoveryComplete FALSE
callSignalAddress
rasAddress
ipAddress :
ip 'CA40F944'H
port 53559
terminalType
mc FALSE
undefinedNode FALSE
gatekeeperIdentifier {"gk1.xxx.com.xx"}
endpointVendor
vendor
t35CountryCode 181
t35Extension 0
manufacturerCode 18
timeToLive 60
keepAlive TRUE
endpointIdentifier {"82691D700000006C"}
willSupplyUUIEs FALSE
maintainConnection TRUE
Feb 11 19:42:38.183: RAS OUTGOING ENCODE BUFFER::= 0E 402CD206 0008914A 00040000 0100CA40 F944D137 00110067 006B0031 002E0069 002D0070 006F0077 00650072 002E0063 006F006D 002E0068 006B00B5 00001228 8F000002 003B0180 211E0038 00320036 00390031 00440037 00300030 00300030 00300030 00300036 00430100 0180
Feb 11 19:42:38.191:
Feb 11 19:42:38.207: RAS INCOMING ENCODE BUFFER::= 12 402CD206 0008914A 00020022 0067006B 0031002E 0069002D 0070006F 00770065 0072002E 0063006F 006D002E 0068006B 1E003800 32003600 39003100 44003700 30003000 30003000 30003000 30003600 430C8802 003B0100
Feb 11 19:42:38.215:
Feb 11 19:42:38.215: RAS INCOMING PDU ::=

Cisco UC 520 FXO port, dual or single line?

I am needing to configure the FXO ports on a UC 520 to accept 2 numbers instead of just 1, is this possible?

timer media-inactive To enable the timer for media inactivity detection using the digital signal processor (DSP) (based on RTP as the only criterion) and to configure a multiplication factor based on the real-time control protocol (RTCP) timer interval, use the timer media-inactive command in gateway configuration mode. To reset to the default, use the no form of this command. timer media-inactive multiple no timer media-inactive multiple
When the timer media-inactive command is used, the gateway uses the inactivity timer as a combination of the timer media-inactive command and the ip rtcp report interval command. The timer media-inactive command uses DSP statistics. This capability is based on the configuration of callfeature parameters using application command-line interface (CLI) to enable control. The media are considered inactive only if there is no transfer of RTP packets in the send direction and no RTP packets in the receive direction. If RTP is present in either the send or receive direction, it is considered active. In this mode, DSP filters out any comfort noise packets, and the presence of any comfort noise packet is considered inactivity in either direction. The multiple argument (or multiplication factor) is multiplied by the interval that is set using the ip rtcp report interval command. This command configures the average interval between successive RTCP report transmissions for a given voice session. For example, if the value argument is set to 25,000 milliseconds, an RTCP report is sent every 25 seconds, on average. If no RTP packets are received during the calculated interval, the call is disconnected. The gateway signals the disconnect to the VoIP network and the time-division multiplexing (TDM) network so that upstream and downstream devices can clear their resources.
Sent from Cisco Technical Support iPad App

Cisco Solution for FXO Disconnect Problem

I have Panasonic PBX-->FXO-->1751V------WAN-----1751V-->FXO--Panasonic PBX. I am facing a serious problem with FXO on both ends becoming busy after some few initial calls.
What is the exact Cisco solution for this FXO disconnect problem? I have read and applied some Netpro tips for similar case but none could help.
Can we have Cisco expert explain this??

Hello, I think this url may be helpful:
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00800ae2d1.shtml
Regards.

Cisco IAD 2432 24 FXS License

wanted to know what license I need to order if I am going to be using this device.
CCM license?

This product is not Small Business class of product, thus your's question doesn't belong here and your's chances to receive valuable answer are rather low.
Please delete it here and recreate in more appropriate forum.

"frags delayed" counter incrementing for Voice PVC

Hi,
We are using VoFR between two Cisco 2610 using FXO\FXS Cards. It is a point-point link with two PVCs, one for Voice and one for Data.
I have implemented Traffic-Shaping and FRF. However when i do a "show frame pvc " command, i can see "frags delayed"counter incrementing for the Voice PVC, indiciating delay in sending packets and thus compromising Voice Quality.
1. Is it normal to have this counter increasing ? What is the acceptable percentage i.e "frags delayed \ total frags" ?
2. Is there anything i can do ? Would PVC Priority Queuing help ?
I need to be sure if PVC Priority is the solution, as we would have to do a Flash Upgrade to install the new software with this feature.
++++++++++++++++++++++++++
show frame pvc 103
PVC Statistics for interface Serial0/0 (Frame Relay DTE)
DLCI = 103, DLCI USAGE = LOCAL, PVC STATUS = ACTIVE, INTERFACE = Serial0/0.3
input pkts 373951 output pkts 374604 in bytes 11542352
out bytes 12245392 dropped pkts 0 in FECN pkts 0
in BECN pkts 0 out FECN pkts 0 out BECN pkts 0
in DE pkts 0 out DE pkts 0
out bcast pkts 5474 out bcast bytes 1571038
pvc create time 11w3d, last time pvc status changed 04:06:31
Service type VoFR-cisco
Voice Queueing Stats: 0/100/0 (size/max/dropped)
Current fair queue configuration:
Discard Dynamic Reserved
threshold queue count queue count
64 16 2
Output queue size 0/max total 600/drops 0
configured voice bandwidth 30000, used voice bandwidth 0
fragment type VoFR-cisco fragment size 320
cir 32000 bc 320 be 0 limit 40 interval 10
mincir 32000 byte increment 40 BECN response no
frags 374604 bytes 12261814 frags delayed 6501 bytes delayed 1609296
shaping inactive
traffic shaping drops 0
+++++++++++++++++++++++++++++++++++

The following links explains the delay in voice traffic and gow to do traffic policing
VoIP over Frame Relay with QoS (Fragmentation, Traffic Shaping, LLQ / IP RTP Priority)
http://www.cisco.com/warp/public/788/voice-qos/voip-ov-fr-qos.html#15
Troubleshooting Output Drops with Priority Queueing
http://www.cisco.com/warp/public/105/priorityqueuedrops.html
Understanding Delay in Packet Voice Networks
http://www.cisco.com/warp/public/788/voip/delay-details.html
Voice QoS: ToS-CoS Mapping Via LLQ
http://www.cisco.com/warp/public/788/voice-qos/tos-cos.html
Frame Relay Traffic Shaping for VoIP and VoFR
http://www.cisco.com/warp/public/788/voip/fr_traffic.html

Connecting FAX to PSTN line using FXS and FXO ports

I have 2 cisco 1760 routers with FXS and FXO card installed in each. I have to transport 2 PSTN lines from Head Office to remote loaction using FXS and FXO cards and these lines will be used for voice calls and FAX( one line each for FAX and voice). I have configured the router for voice lines and it is working fine. Now I am using the same config for running fax machine but it is not working. Can anyone help me out how to configure FAX in this scenario also if anyone can share any sample config. I am attaching my config for both routers. Right now both routers are connected with a cross-over cable for lab test but we will connect them later using satellite connection.
HO Router (FXO card)
Current configuration : 1718 bytes
version 12.3
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router
boot-start-marker
boot-end-marker
enable password cisco
no aaa new-model
resource policy
mmi polling-interval 60
no mmi auto-configure
no mmi pvc
mmi snmp-timeout 180
voice-card 2
voice-card 3
ip subnet-zero
ip cef
no ip dhcp use vrf connected
no ip domain lookup
no ftp-server write-enable
voice class codec 10
interface FastEthernet0/0
ip address 10.10.10.1 255.255.255.248
speed auto
h323-gateway voip bind srcaddr 10.10.10.1
ip classless
no ip http server
control-plane
voice-port 2/0
output attenuation 0
echo-cancel coverage 32
no vad
no comfort-noise
timeouts interdigit 3
timeouts call-disconnect 3
connection plar opx 2001
description Remote PSTN#:35296913
music-threshold -70
voice-port 2/1
output attenuation 0
echo-cancel coverage 32
no vad
no comfort-noise
timeouts interdigit 3
timeouts call-disconnect 3
connection plar opx 2002
description Remote PSTN#:35296914
music-threshold -70
voice-port 3/0
voice-port 3/1
dial-peer voice 2000 voip
destination-pattern 200.
no modem passthrough
voice-class codec 10
session target ipv4:10.10.10.2
incoming called-number .
dtmf-relay cisco-rtp h245-signal h245-alphanumeric
fax-relay ecm disable
fax rate 7200
fax nsf 000000
no vad
dial-peer voice 1321 pots
description line 1
huntstop
destination-pattern 1321
port 2/0
dial-peer voice 1322 pots
description line 2
huntstop
destination-pattern 1322
port 2/1
line con 0
password cisco
line aux 0
line vty 0 4
password cisco
login
end
Remote Router (FXS Card):
version 12.3
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router
boot-start-marker
boot-end-marker
enable password cisco
mmi polling-interval 60
no mmi auto-configure
no mmi pvc
mmi snmp-timeout 180
voice-card 2
voice-card 3
no aaa new-model
ip subnet-zero
ip cef
no ip domain lookup
no ftp-server write-enable
voice class codec 10
interface FastEthernet0/0
ip address 10.10.10.2 255.255.255.248
speed auto
h323-gateway voip bind srcaddr 10.10.10.2
interface Ethernet1/0
no ip address
shutdown
half-duplex
ip classless
no ip http server
control-plane
voice-port 2/0
description PSTN#:
voice-port 2/1
description PSTN#:
voice-port 3/0
voice-port 3/1
dial-peer voice 2001 pots
description Remote
huntstop
destination-pattern 2001
port 2/0
dial-peer voice 2002 pots
description Remote
huntstop
destination-pattern 2002
port 2/1
dial-peer voice 1320 voip
destination-pattern 132.
no modem passthrough
voice-class codec 10
session target ipv4:10.10.10.1
incoming called-number .
dtmf-relay cisco-rtp h245-signal h245-alphanumeric
fax-relay ecm disable
fax rate 7200
fax nsf 000000
no vad
line con 0
password cisco
line aux 0
line vty 0 4
password cisco
login
end

In your voice class codec 10 there aren't any codecs declared.
Add G.711 codec in this way:
voice class codec 10
codec preference 1 g711alaw
If the fax communication fails again try to disable T.38 and try fax passthrough mode:
no fax rate
modem passthrough nse codec g711alaw
fax protocol pass-through g711alaw
Regards.

FXS and FXO configuration and design help

hi all...
i am new to voip configuration... and i have to configure the voip for our branch office... we have 2610 with the 2FXS and 2FXO prot... now i have one pulblic IP address which is used by my local lan user...and we have configured NAT/PAT on our VPN concerntrator... now we are looking to establish the VOIP phone (analogphone with the help of FXO and FXS) we have ASTRIK server at our main office and i want to register my local office analog phone with server at our main office... now what kind of configuration we need on our 2610 in order to configure voip...
connectivity:
ADSL connection form ISP---VPN 3005---D-Link nonmanagable swithch---LAN
at present we have above connectivity and now i want to add my 2610 router with analogphone connected to it... how can i connect and how can i configure it...?
regards
Devang

you need to configure pots and voip dialPeers for the incoming/outgoing legs of the calls.
pots dialPeers will be used for the FXO/FXS ports. voip dialPeers will be used for connection to ASTERISK voip pbx.
you need voice port configuration for your FXS to connect to your analogPhone.
you need voice port configuration for your FXO to connect to analog pstn. (if you have any)
see these links for more info:
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010ae1c.shtml
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080147524.shtml
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example09186a00801bc341.shtml
http://www.cisco.com/en/US/tech/tk652/tk653/tech_configuration_examples_list.html
there are plenty of examples throughout these for reference.

Cisco 2610xm NM2V FXO FXS

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