Direct IP Dialing

Hi Everyone,
I am trying to configure IP dialing but two linksys PAP 2T ATA one in ksa and the other in india, i have the configured the line 1 of pap2T to dial to line 2 of PAP 2T in india. I have done the settings as metioned in the document "Configuring IP Dialing on the PAP2 document on the linksys knowledge Base".
I would be glad if any one could tell me as what values i have to provide for the following fields
1) PROXY
2) USER ID
3) PASSWORD
In the document provided by linksys the above fields are left as it is intended to call between ATA on the same network which is not i am looking for,Look forward to have any inputs on the issue.
Thanks in Advance.

Your remarks are correct about the voip provider. When an adapter "registers" with a voip provider's proxy the adapter initiates conversation and the exchange removes some of the NAT router problems that you can encounter when you are using direct ip calling. The NAT Keep Alive function will send something every 15-seconds to the proxy and will further reduce problems with your NAT router. The registration, of course, also gives the voip provider your current ip address to send an incoming call. Frequent registration every few minutes keeps your ip address current. It is possible, though, to use direct ip calling without using a voip provider.
I believe from what you have outlined that your adapter is sending the call request to the distant ata and the odds are the packet is not being received by the distant ata. Running a sip debug trace, first on the sending ata will establish that the sip invite is being sent. Then running a sip debug trace on the receiving ata will establish that the sip invite is not being received. Or I could be mistaken and the traces will show error messages flowing back and forth. In any event it is useful to know what is happening.
To run a sip debug trace you need to download and install a syslog program on a pc. Usually the pc is on a network local to the ata because you don't need to worry about the syslog packets being blocked by a router. In any event, you put the pc's ip address on the PAP's system tab under Debug Server, and you set the Debug Level to 3 on the system tab. On the Line Tab you set the Sip Debug Option to FULL. Then you run a call. On the pc running the syslog program, the syslog messages will display and be saved to the pc's hard drive.
If you do not have a syslog program, send me a private message with your email address and I will send you a syslog program that will run on a Windows pc.
In your dial plan the SO should be S0 (S zero). This is not causing the problem, though.
Message Edited by hw on 08-02-2009 10:15 PM

Similar Messages

Direct Outward Dial on CME 3.1

I am at a complete loss. Cisco want 800 USD to help me.
I need to setup Direct Outward Dial on my CME 3.1 (3725). I need to have one of my two POTS lines show up on the telephones mapped to a button.
I have searched for quite a while, and tried most everything I can think of. I have failed.
Any help is appreciated!

It is not clear to me what you are trying to accomplish, but it sounds like a scenerio I deploy on a regular basis.
If you want to force a particular DN to use a specific analog fxo line for outbound calls, simply create a translation pattern and apply it to the ephone-dn. Basically, you will be prepending a '1#' to the dialed phone number and have a specific dial-peer matching 1# using the fxo voice port as a destination. For example:
translation-rule 1
Rule 0 ^.* 1#
translation-rule 2
Rule 0 ^.* 2#
dial-peer voice 100 pots
destination-pattern 1#0
port 1/0/0
forward-digits 1
dial-peer voice 101 pots
destination-pattern 1#1..........
port 1/0/0
forward-digits 11
dial-peer voice 102 pots
destination-pattern 1#[2-9]......
port 1/0/0
forward-digits 7
dial-peer voice 103 pots
destination-pattern 1#[4,9]11
port 1/0/0
forward-digits 3
dial-peer voice 200 pots
destination-pattern 2#0
port 1/0/1
forward-digits 1
dial-peer voice 201 pots
destination-pattern 2#1..........
fax rate disable
port 1/0/1
forward-digits 11
dial-peer voice 202 pots
destination-pattern 2#[2-9]......
port 1/0/1
forward-digits 7
dial-peer voice 203 pots
destination-pattern 2#[4,9]11
port 1/0/1
forward-digits 3
ephone-dn 1
number 1001
label Line 1
description 555-1212
name Line 1
call-forward busy 7000
call-forward noan 7000 timeout 15
translate called 1
hold-alert 120 idle
ephone-dn 2
number 1002
label Line 2
description 555-1213
name Line 2
call-forward busy 7000
call-forward noan 7000 timeout 15
translate called 2
hold-alert 120 shared
In addition, when you do this you usually want to do the same in reverse: a specific fxo port to ring a specific line when an inbound call comes in, you do this with the connection plar-opx command on the fxo voice port, for example:
voice-port 1/0/0
input gain 10
no comfort-noise
connection plar opx 1001
caller-id enable type 1
voice-port 1/0/1
input gain 10
no comfort-noise
connection plar opx 1002
caller-id enable type 1
Let me know if I'm on the right track...
/Rick

Direct inward dialing

When direct inward dialing the TFN routes to agent even if agent is in busy status, can it be set to only send calls to agent on available status and to agent's VM when not?

The only one that is simple is the option under controls and restrictions that can 'send the call directly to voicemail if the user is logged off'
If you want to also do the Available one, then you use web service from inside a campaign to check the agents current status.

How can I do direct ip dialing?

Hello:
Usually I configure the dial peers with the command session target with the remote gw ip address.
there is any way to make the call using an ip from the caller endpoint?
In each call the user can dial a different ip address and I need that the gw sends the setup to that ip.
If I can not do it with ios, there is some product from cisco that it can do it?
Thanks everyone

You can simply extract the dry signal from the project file.
When you ctrl-click or right-click your project (.band file) and use the command "Show package contents" from the contextual menu, you will see the internal folders like this:
The "Media" folder contains the dry recordings as aiff files. The screenshot above is from GarageBand 10.0. In GarageBand '11 the aiff files will be directly in the Media folder.

ESS Who's Who (ECC6, WD) - Error with telephone direct dial link

Hi there -
We are on ECC6, NW 2004s (SP14). I have the following field in our Who's Who 'Output field detail' -
SYHR_A_P0105_AF_TEL_NR - Area code + local access code
Also in config, I have the checkbox DESELECTED for 'Allow direct telephone dialing.'
Issue with Who's Who service:
The telephone number is displayed as a hyperlink. When I click the link, I get the following message: 'Search resulted more than 1500 records, Only first 1500 records is displayed.'
Then if I click the link again, I get a dump: 500 Internal Error,
com.sap.tc.webdynpro.modelimpl.dynamicrfc.WDDynamicRFCExecuteException: Specify a server name , error key: RFC_ERROR_SYSTEM_FAILURE
Does anyone know how to get rid of the link - direct dialing is not a requirement for my client. Also - the first error message makes no sense. Any help would be greatly appreciated - I am stumped.
Regards,
Karen

a quick note from a call we opened this week
...More significantly is that SAP_ESS is only one part of
the ESS ERP2005 product - if you look at the stack
schedule the next stack is due out Week 48 - last
week of November 2007...

Site to Site calling issue - Cisco 2911 Dial Peer Configuration

My customer dials from remote site to main site to their main site number, the call by-passes their auto attendant and goes directly to any random available party.
At first fingers were pointing to the their PBX, however we noticed one of their sites that wasn't managed by our company did not have the issue. We cut that site over to our service and the issue started right up. I believe it is possibly due to the way the dial peers are configured and how the calls route into the PBX. Unfortunately I do not understand much about them and curious to know if anyone has any history on a issue similiar to this or any input whatsoever?
Cisco equipment/Dialpeer config below ........
co IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version 15.2(4)M4, RELEASE SOFTWARE (fc2) - Cisco CISCO2911/K9
dial-peer voice 100 voip
description --- VoIP Dial-Peer ---
translation-profile outgoing 7digit
huntstop
preference 1
service session
destination-pattern .T
progress_ind setup enable 3
session protocol sipv2
session target sip-server
incoming called-number .T
voice-class codec 99
dtmf-relay rtp-nte
fax-relay ecm disable
fax rate 14400
fax nsf 000000
ip qos dscp af41 signaling
no vad
dial-peer voice 150 voip
permission none
description 900 block
huntstop
destination-pattern 1900T
session protocol sipv2
session target sip-server
voice-class codec 99
dtmf-relay rtp-nte
ip qos dscp af41 signaling
no vad
dial-peer voice 151 voip
permission none
description 900 block
huntstop
destination-pattern 900T
session protocol sipv2
session target sip-server
voice-class codec 99
dtmf-relay rtp-nte
ip qos dscp af41 signaling
no vad
dial-peer voice 101 pots
description --- INCOMING Calls from PBX ---
incoming called-number .T
direct-inward-dial
dial-peer voice 1001 pots
description --- Calls to the PBX ---
preference 3
destination-pattern .T
port 0/0/1:23
forward-digits 4
Here is some ISDN debug information
BAD CALL
Protocol Profile = Networking Extensions
0xA11C0201420201008014484152545F20484F54454C535F434C4159544F4E
Component = Invoke component
Invoke Id = 66
Operation = CallingName
Name Presentation Allowed Extended
Name = XXXXXXXXXXX
Display i = ''XXXXXXXXXXX''
Calling Party Number i = 0x2180, ''XXXXXXXXXX''
Plan:ISDN, Type:National
Called Party Number i = 0x80, ''6551''
Plan:Unknown, Type:Unknown
Aug 19 16:10:47.242 GMT: ISDN Se0/0/1:23 Q931: RX <- ALERTING pd = 8 callref = 0xAB15
Channel ID i = 0xA98381
Exclusive, Channel 1
Aug 19 16:11:02.634 GMT: ISDN Se0/0/1:23 Q931: RX <- CONNECT pd = 8 callref = 0xAB15
Channel ID i = 0xA98381
Exclusive, Channel 1
Aug 19 16:11:02.634 GMT: ISDN Se0/0/1:23 Q931: TX -> CONNECT_ACK pd = 8 callref = 0x2B15
GOOD CALL
Protocol Profile = Networking Extensions
0xA116020144020100800E475245454E204D4F554E5441494E
Component = Invoke component
Invoke Id = 68
Operation = CallingName
Name Presentation Allowed Extended
Name = XXXXXXXXXXXXXXXXXX
Display i = ''XXXXXXXXXXX''
Calling Party Number i = 0x2180, ''XXXXXXXXXX''
Plan:ISDN, Type:National
Called Party Number i = 0x80, 'XXXX''
Plan:Unknown, Type:Unknown
Aug 19 16:15:07.999 GMT: ISDN Se0/0/1:23 Q931: RX <- ALERTING pd = 8 callref = 0xAB17
Channel ID i = 0xA98381
Exclusive, Channel 1

I done the configration via CCA and the running conf i can see two voip dial peer. this is the site where all trunk line roured. Customer from other site2 needs to call outside by taking line from site1.
dial-peer voice 2100 voip
corlist incoming call-internal
description **CCA*INTERSITE inbound call to SITE 1
translation-profile incoming multisiteInbound
incoming called-number 82...
voice-class h323 1
dtmf-relay h245-alphanumeric
fax protocol cisco
no vad
dial-peer voice 2101 voip
corlist incoming call-internal
description **CCA*INTERSITE outbound calls to SITE2
translation-profile outgoing multisiteOutbound
destination-pattern 81...
session target ipv4:192.168.50.1
voice-class h323 1
dtmf-relay h245-alphanumeric
fax protocol cisco
no vad
no dial-peer outbound status-check pots

FAX and Dial-access with 5350

I have a 5350 router, and I get some troubles. I can't use fax over wan links. Before I had a 5300 router and it works fine (same IOS version). Voice works fine, but I can't use fax. This interface is E1 with ISDN Q-Sig.
Configuration below...
==
version 12.2
service timestamps debug datetime msec localtime show-timezone
service timestamps log datetime msec localtime show-timezone
service password-encryption
hostname 5350
no boot startup-test
logging buffered 20480 debugging
aaa new-model
aaa authentication login default local
aaa authentication enable default enable
aaa session-id common
resource-pool disable
calltracker enable
spe default-firmware spe-firmware-1
ip subnet-zero
ip cef
frame-relay switching
isdn switch-type primary-qsig
isdn voice-call-failure 0
voice call send-alert
voice call carrier capacity active
voice rtp send-recv
voice service voip
fax protocol t38 ls-redundancy 0 hs-redundancy 0
mta receive maximum-recipients 0
controller E1 3/0
pri-group timeslots 1-31
interface FastEthernet0/0
ip address x.x.x.x x.x.x.x
duplex auto
speed auto
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
interface Serial3/0:15
no ip address
isdn switch-type primary-qsig
isdn overlap-receiving
isdn incoming-voice modem
isdn bchan-number-order ascending
isdn sending-complete
no cdp enable
interface Group-Async0
ip unnumbered FastEthernet0/0
encapsulation ppp
ip tcp header-compression
async mode dedicated
peer default ip address pool mypool
ppp authentication pap
group-range 1/00 1/59
ip local pool mypool x.x.x.x x.x.x.x
ip classless
ip route 0.0.0.0 0.0.0.0 x.x.x.x
no ip http server
call rsvp-sync
voice-port 3/0:D
bearer-cap Speech
voice-port 3/1:0
compand-type a-law
mgcp profile default
dial-peer cor custom
dial-peer voice 8400 pots
application data_dialpeer
incoming called-number 8400
port 3/1:0
forward-digits all
dial-peer voice 1000 pots
destination-pattern 1...
direct-inward-dial
port 3/0:D
forward-digits all
dial-peer voice 1001 pots
destination-pattern 8...
direct-inward-dial
port 3/0:D
forward-digits all
dial-peer voice 2000 voip
destination-pattern 12..
session target ipv4:x.x.x.x
dtmf-relay rtp-nte
ip qos dscp cs5 media
5350#sh ver
Cisco Internetwork Operating System Software
IOS (tm) 5350 Software (C5350-IS-M), Version 12.2(11)T8, RELEASE SOFTWARE (fc1)
TAC Support: http://www.cisco.com/tac
Copyright (c) 1986-2003 by cisco Systems, Inc.
Compiled Thu 27-Mar-03 22:32 by hqluong
Image text-base: 0x60008948, data-base: 0x61380000
ROM: System Bootstrap, Version 12.2(1r)1, RELEASE SOFTWARE (fc1)
BOOTLDR: 5350 Software (C5350-BOOT-M), Experimental Version 12.1(20000922:142008) [nag-flo_t_0110 101]
cfw-5350-voip uptime is 1 day, 3 hours, 8 minutes
System returned to ROM by reload at 00:23:01 brz Fri Jan 21 2000
System image file is "flash:c5350-is-mz.122-11.T8.bin"
cisco AS5350 (R7K) processor (revision T) with 131072K/65536K bytes of memory.
Processor board ID JAE070401D5
R7000 CPU at 250Mhz, Implementation 39, Rev 1.0, 256KB L2, 2048KB L3 Cache
Last reset from IOS reload
Channelized E1, Version 1.0.
Bridging software.
X.25 software, Version 3.0.0.
SuperLAT software (copyright 1990 by Meridian Technology Corp).
Primary Rate ISDN software, Version 1.1.
Manufacture Cookie Info:
EEPROM Type 0x0001, EEPROM Version 0x01, Board ID 0x32,
Board Hardware Version 3.34, Item Number 800-5171-02,
Board Revision C0, Serial Number JAE070401D5,
PLD/ISP Version 2.2, Manufacture Date 24-Jan-2003.
Processor 0x14, MAC Address 0x0B5FDAB22
Backplane HW Revision 1.0, Flash Type 5V
2 FastEthernet/IEEE 802.3 interface(s)
37 Serial network interface(s)
60 terminal line(s)
2 Channelized E1/PRI port(s)
512K bytes of non-volatile configuration memory.
32768K bytes of processor board System flash (Read/Write)
8192K bytes of processor board Boot flash (Read/Write)
Configuration register is 0x2102
5350#
===
Can someone help me?
regards

try this:
voice service pots
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
voice service voip
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
dial-peer voice 900 voip
permission term
description your outgoing dialpeer
destination-pattern ^1234..
modem relay nse codec g711alaw redundancy
modem relay latency 150
modem relay sprt retries 6
fax-relay ecm disable
fax nsf 000000
fax protocol t38 ls-redundancy 2 hs-redundancy 2 fallback pass-through g711ulaw
no vad
and make sure you have g711alaw and g711ulaw in your codec preference list.

NOT RECEIVING THE COMPLETE DIALED STRING FROM AN ISDN TRUNK PBX - AS5850

Hi
The problem i'm facing is the next:
I have a Cisco AS5850 with an Euro ISDN trunk connected to a PBX. The pbx send out the calls in order the A5850 terminate these calls in a remote gateway. I receive only a maximum of 12 digits of the complete string dialed in a phone hooked up in the PBX side.
Is it normal to receive only a max of 12 digits or there is something to do to receive as much digits as those come from the PBX??
My dial-peer looks like this:
dial-peer voice 13 pots
incoming called-number 0T
destination-pattern 0T
direct-inward-dial
port 2/7:D
dial-peer voice 14 voip
destination-pattern 0T
session target ipv4:209.210.174.17
tech-prefix 1051
fax rate 9600
fax protocol t38 ls-redundancy 2 hs-redundancy 2 fallback none
and the output from the debug dialpeer is this:
1w4d: destination pattn: 1234567891 expanded string: 1234567891
1w4d: MatchNextPeer: Peer 7 matched
1w4d: MatchNextPeer: Peer 8 matched
1w4d: dpMatchPeersMoreArg: Result=0 after MATCH_ORIGINATE
Thanks in advance for your help!
Oscar

Do a debug isdn q931 and see what comes in from the PBX. Post it here so we can see. They could be sending the digits using overlap sending , so the intial setup may not have all the digits that you want or need.
Also, your VOIP and POTS dial peers have the same destination pattern. There is nothing stopping an inwards ISDN call from being hairpinned back out the PRI if it has a leading 0. You might want to make your dial plan a bit more specific.

IP address based dial-peer matching

Hi,
is it possible to define a rule for incoming voip dial-peer matching based on the source IP address of originating VOIP gateway ?
I need it for differentiate outgoing POTS lines based on the source gateway. Any ideas how to do it ?
Thanks,
Vladimir

HI
Yes is it, to using voice source-group features..
Simple examble:
voice source-group vladimir
access-list 50
disconnect-cause call-reject
translation-profile incoming 50
voice translation-profile 50
translate called 50
voice translation-rule 50
rule 1 /^1$.*$/ /6501\1/
access-list 50 permit 10.0.0.0 0.0.0.255
dial-peer voice 52 pots
destination-pattern 650T
progress_ind alert enable 8
progress_ind progress enable 8
direct-inward-dial
port 7/0:D
rgds,
Ismo

Cisco dial-peer path selection with "preference"

Hi everybody,
for a test lab environment i'm testing the integration between cisco voice gateway 3925 and third party voice gateway by means of isdn PRI.
here the connection schema:
PSTN (emulated)-----> port0/0/0-Cisco3925-port0/0/1 <------- Third party Voice Gateway
                                                              |     (ethernet)
                                                      Cisco CUCM (172.23.112.20)
in brief:
- i'm emulating PSTN with a cisco voice gateway, this gateway is connected to cisco3925's port 0/0/0.
- cisco3925's port 0/0/1 is connected to Third party Voice Gateway.
- cisco 3925 speaks with Cisco CUCM in H323.
Now let's go for an incoming call from the PSTN when 3925 has no connection to CUCM, with called number 321672711 (321672... is the GNR of the site):
1. inbound: dial-peer 110 finds match so the called number is transformed to 591711 (it is a DN not registered to SRST cisco gateway)
2. outbound: i expect dial-peer 100 to be matched, because 172.23.112.20 is no more reacheable. From the show call active voice dial-peer 1 is matched as the attached. I need to set preference 1 in dial-peer 100 because when WAN is UP i don't want dial-peer 100 to be matched (and it works). But when WAN is down dial-peer 100 must match. If i remove preference 1, dial-peer 100 finds match; but for correct path selection i cannot remove it.
What am I forgetting?
thanks for support
voice translation-rule 1
rule 1 /^321672/ /591/
voice translation-profile ENTRANTE
translate called 1
(translate calling omitted)
dial-peer voice 1 voip
description Inbound per USCENTI - Outbound per ENTRANTI
corlist incoming CSSSRSTInternazionali
tone ringback alert-no-PI
destination-pattern 591...
session target ipv4:172.23.112.20
voice-class codec 1
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 100 pots
preference 1
translation-profile outgoing NOMIG
destination-pattern 591...
port 0/0/1:15
dial-peer voice 110 pots
corlist incoming CSSSRSTInternazionali
description Inbound per ENTRANTI
translation-profile incoming ENTRANTE
incoming called-number 321672...
direct-inward-dial
port 0/0/0:15

Hello Marco,
There could be two possibilities:
1. To avoid dial-peer 1 being selected in the dialplan match, when gateway is trying to route the call, you can configure ICMP Probe , which would mark dial-peer as down, in case of WAN failure. So call will use dial-peer 100, automatically, as that will only be an possible match.
Here is document , in case you are interested in ICMP Probe:
http://www.cisco.com/c/en/us/td/docs/ios/voice/command/reference/vr_book/vr_m3.html#wp1397581
2. Ideally default dial-peer hunting mechanism is, Longest - Preference - Random , so as both the dial-peer has same destination pattern, in terms of specific digits and number of wild cards. So it should be looking as preference value of two possible matches, so in this test dial-peer 1 would win. Router will try to route the call using that dial-peer, if fails it should automatically fall back to dial-peer 100 as next choice.
But please note that it will still use dial-peer 1 at first attempt, as dial-peer status is not linked to interface status or WAN status. To verify this theory , you can remove session target command, and you will see that dial-peer 1, is not even selected in match, that's because removing session target command, will mark is as DOWN for outgoing status.
Taking below said debugs would help further, in case configuring ICMP probe is not viable option.
debug voip ccapi inout ( it will help understand , dial-peer match and hunting process ).
debug voip dialpeer inout
Hope that helps.

UC560 T1E1 with Direct TCP SIP

Hi,
Firstly I am new to cisco's voice setup, and have a UC560-T1E1-K9 which I need to configure as just a basic gateway. Essentially i want to setup trunk-to-trunk between ISDN PRI and TCP SIP but do not have the first idea where to start.
The main aim is to receive calls over ISDN PRI (E1 NET5) and pass the call stright through to a direct tcp sip trunk (incoming only).
Then configure FXS for a few numbers for two way calling.
Hope someone can give me a couple of pointers as the CUE is not very helpful
Thanks,
Alex.

Hi,
You have two approaches.
Option A
Use CCA to configure the unit from scratch. Configure the unit for your ISDN connection and test using the fxs ports. Manually configure a SIP dial-peer to route inbound PSTN calls to your Lync server. You'll need to tweak the configuration at various places to make it work. You therefore need to be pretty confident in understanding all the work CCA has done for you. The benefit this, is you have a local outbound dial-plan pre-configured on the unit.
Option B
Configure the unit yourself from scratch. Rough outline:
1. Configure E1 controller for voice
2. Setup the DSP modules for your region
3. configure a pots inbound dial peer to accept calls from the PSTN
4. configure an outbound SIP dial-peer to send the calls to Lync.
Lync expects calls to be prefixed with a + & E.164 and bare in mind the telco may only send you the last 6 sigits of the called number
Here's an example for UK ISDN with 30 timeslots: note module slots will be incorrect in this code, example is generic IOS and not specific to UC560.
1. Configure E1 controller for voice
card type e1 0
network-clock-participate wic 0 ! something like this, you may need to work out which slot you have
network-clock-participate slot 1 ! or something like this
network-clock-select 5 E1 1/0 ! you get the idea....
controller E1 1/0
pri-group timeslots 1-31
interface Serial1/0:15 ! do a show run to check the right interface slot number
description "use dial-peer voice 10 to reach me for outbound"
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn timer T309 400000
isdn incoming-voice modem
isdn T309-enable
isdn send-alerting
isdn sending-complete
no cdp enable
Your E1 card should now be up
c2821.1.hex#show controllers e1
E1 1/0 is up.
Applique type is Channelized E1 - balanced
Cablelength is Unknown
No alarms detected.
alarm-trigger is not set
Version info Firmware: 20090408, FPGA: 13, spm_count = 0
Framing is NO-CRC4, Line Code is HDB3, Clock Source is Line.
CRC Threshold is 320. Reported from firmware is 320.
Data in current interval (652 seconds elapsed):
     0 Line Code Violations, 0 Path Code Violations
     0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
     0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs
Total Data (last 24 hours)
     0 Line Code Violations, 0 Path Code Violations,
     0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins,
     0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs
2. Setup the DSP modules for your region
voice-port 1/0:15
cptone GB
bearer-cap Speech
3. configure a pots inbound dial peer to accept calls from the PSTN
I suggest at this point, you debug isdn q931 and dial in. See how many digits the telco sends you.
Configure a number translation to put this number in to E.164
In my example we take the 6 digits the telco sends us and prepend 441234
voice translation-rule 10
rule 1 /^$.+$$/ /441234\1/
voice translation-profile incomingisdn
translate called 10
in exec mode:
SOV_TAG1#test voice translation-rule 10 567890
Matched with rule 1
Original number: 567890 Translated number: 441234567890
Original number type: none      Translated number type: none
Original number plan: none      Translated number plan: none
dial-peer voice 1000 pots
description Inbound POTS dial-peer
translation-profile incoming incomingisdn
incoming called-number .+
direct-inward-dial
port 1/0:15
4. configure an outbound SIP dial-peer to send the calls to Lync. Lync expects calls to be prefixed with a + & E.164 so we're going to add a plus
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
codec preference 4 g729br8
voice translation-rule 20
rule 1 /$.+$/ /+\1/
voice translation-profile addaplus
translate called 20
in exec mode
#test voice translation-rule 20 123
Matched with rule 1
Original number: 123    Translated number: +123
dial-peer voice 1010 voip
description SIP Trunk to Lync
translation-profile outgoing addaplus
preference 5
destination-pattern ^441234.+ ! note this is the area code we added to make the number into E164
voice-class codec 1
voice-class sip outbound-proxy ipv4:192.168.10.10 ! ip address of lync
session target dns:lync.mydomain.net ! your lync realm
session protocol sipv2
session transport tcp
dtmf-relay rtp-nte
no vad
and that's about it apart from getting the fx0 ports sorted - one step at a time.
For outbound you added an incoming SIP dial-peer and an outgoing pots dial peer.
Adam
VoIP.co.uk

FXO not giving the dial tone

Hi,
We have FXO ports on the router and SIP trunk towards ITSP. People used to dial into FXO and get the dial tone to callout using the SIP trunk. But after the upgrade of the IOS, this functionality has stopped. Now if you call the FXO, you get busy tone. The IOS has been upgraded from 12.4(11) to 15.1(M4) to basically its a big leap.
I strongly believe that we need to make some configuration to make it work like before.
Please advise.
Attachached are the logs from the "debug vpm all"

[+] for Calro
Here is the new feature complied in 15.X release
http://www.cisco.com/en/US/docs/ios/15_1/release/notes/151-2TNEWF.html
Toll Fraud Prevention
In Cisco IOS Release 15.1(2)T, the Toll Fraud Prevention feature is supported as below:
•Source IP address authentication is enabled on incoming IPv4 H323/ or SIP trunk calls. The source IP address of any incoming IPv4 H323 or SIP trunk calls will be authenticated based on:
–Manually configured IP address trusted list.
–VoIP dial-peer session target (the state of a VoIP dial-peer must be in "Operation State = UP")
Incoming IPv4 H323 or SIP trunk calls will be rejected if the authentication fails and the default cause-code call-reject (21) disconnects the call.
Execute the show ip address trusted list command to display IP address trusted data and a list of valid source IP addresses. The default behavior can be disabled as shown in the example below:
voice service voip
no ip address trusted authenticate
•Secondary dial-tone is disabled for a call initiated from a FXO port. No secondary dial-tone causes the outgoing call setup to fail if the called number is NULL. The default behavior can be disabled as shown below:
voice-port
secondary dialtone
•Direct-inward-dial is enabled to prevent the toll fraud for incoming ISDN calls. Two-stage dialing is disabled for incoming ISDN calls by default. The incoming called number will then be used for outgoing call setup. The default behavior can be disabled as shown in the example below:
voice service pots
no direct-inward-dial isdn
For more information, see the Cisco Unified Communications Manager Express System Administrator Guide at the following URL:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeadm.
html
Br,
Nadeem
Please rate all useful post.

Need configuration help on producing dial tone

Hello Experts,
I have a Cisco 2921 router with VWIC3-2MFT-T1/E1 card. On this card we have T1-CAS digital line connected. We have been provided with a set of DID numbers. We have a requirement where, when we dial a DID, the router should provide a dial tone, and should allow the user to dial to extension numbers. Not sure if this is feasible. If at all possible, will need to some configuration help.
Thanks
Arabinda

Sure it's possible. What's the T1 connected to? The router will offer two-stage dialing (aka dial tone) when the incoming POTS dial-peer does not have the 'direct-inward-dial' command on it. The router will accept any input and search for an outbound dial-peer (or ephone-dn for locally registered DNs) to match. Be careful if the T1 is connected to the PSTN as this is a toll fraud risk. You need to use CoR to reign in what outbound dial-peers are available to it.
Dial Peer Basics:
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010ae1c.shtml
Class of Restrictions:
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example09186a008019d649.shtml
Please remember to rate helpful responses and identify helpful or correct answers.

2 Stage Dialing with ISDN-30

I have 2 problems here, i have setup a cisco 3640 router with ISDN-30 (E1 Link) as the voice gateway and a SIP server. I have enblock a range of numbers from 62991000 to 62991099 with DID service.
The problems,
1) Whenever I dial any of the numbers above, i will hear a tone and i need to key in the last 4 digits again in order for the call to get through. Can i configure such that the router will pass the called number to SIP server.
2) How can i configure the router so that the reicipent will see my direct number instead of my main line number?
Below is the configuration on the router.
Port 3/0:15 is the ISDN port.
! Last configuration change at 15:55:00 Tue May 17 2005
version 12.3
voice-card 3
ip cef
no ip domain lookup
ip ssh break-string
isdn switch-type primary-net5
no scripting tcl init
no scripting tcl encdir
voice call carrier capacity active
voice rtp send-recv
voice service voip
sip
voice class codec 1
codec preference 1 g729r8
codec preference 2 g729br8
codec preference 3 g723r63
codec preference 4 g723r53
codec preference 5 g711ulaw
codec preference 6 g711alaw
codec preference 7 g728
voice class codec 2
codec preference 1 g711ulaw
no voice hpi capture buffer
no voice hpi capture destination
controller E1 3/0
pri-group timeslots 1-31
interface FastEthernet0/0
ip address x.x.x.x x.x.x.x
ip nat outside
duplex auto
speed auto
interface Serial3/0:15
no ip address
no logging event link-status
isdn switch-type primary-net5
isdn incoming-voice voice
isdn bchan-number-order ascending
no cdp enable
ip nat inside source list 1 interface FastEthernet0/0 overload
no ip http server
no ip http secure-server
ip flow-export destination x.x.x.x
ip classless
ip route x.x.x.x x.x.x.x x.x.x.x
ip route x.x.x.x x.x.x.x x.x.x.x
access-list 1 remark test LAN IP Range
access-list 1 permit x.x.x.x x.x.x.x
voice-port 2/0/0
supervisory disconnect dualtone mid-call
input gain 9
output attenuation 0
cptone SG
timeouts call-disconnect 3
timeouts wait-release 3
shutdown
voice-port 2/0/1
supervisory disconnect dualtone mid-call
input gain 9
output attenuation 0
cptone SG
timeouts call-disconnect 3
timeouts wait-release 3
shutdown
voice-port 2/1/0
shutdown
voice-port 2/1/1
shutdown
voice-port 3/0:15
dial-peer cor custom
dial-peer voice 101 pots
shutdown
destination-pattern 65........$
port 2/0/0
prefix 9
dial-peer voice 102 pots
shutdown
destination-pattern 8...$
port 2/0/0
prefix 8
dial-peer voice 201 pots
preference 1
shutdown
destination-pattern 65........$
port 2/0/1
prefix 9
dial-peer voice 202 pots
preference 1
shutdown
destination-pattern 8...$
port 2/0/1
prefix 8
dial-peer voice 301 voip
incoming called-number 65........$
shutdown
voice-class codec 1
session protocol sipv2
session target ipv4:x.x.x.x
dial-peer voice 302 voip
incoming called-number 7...$
destination-pattern 7...$
voice-class codec 1
session protocol sipv2
session target ipv4:x.x.x.x
dial-peer voice 401 voip
incoming called-number 7...$
shutdown
destination-pattern 7...$
voice-class codec 1
session protocol sipv2
session target ipv4:x.x.x.x
clid network-number 1111
dial-peer voice 402 voip
incoming called-number 7...$
shutdown
destination-pattern 7...$
voice-class codec 1
session protocol sipv2
session target ipv4:x.x.x.x
clid network-number 1111
dial-peer voice 1000 pots
destination-pattern 65........
port 3/0:15
dial-peer voice 1001 voip
incoming called-number 65........
destination-pattern 65........
voice-class codec 1
session protocol sipv2
session target ipv4:x.x.x.x
session transport udp
sip-ua
end

Put the direct-inward-dial command on the pots dial-peer. This will forward the dialed number (with the matched destination-patten stripped) to your SIP server directly and shall not give you a secondary dial-tone.

Dialers from the outside hear the dial tone

Our gateway is connected to the PSTN with a PRI line. Eveything works fine from the inside. We can call outside but when the outside calls us from these line they hear a dial-tone. Why can this happen?

A key concept is that a single phone call through a voice gateway has two call legs, one inbound and one outbound, and each leg matches a dial-peer.
Outbound are the easiest to understand because they operate alot like route statements do to the IP data world; they tell the call/packet where to go. But inbound are important, too, because they instruct the VG on how to receive calls. Alot of VG get by using the (hidden) default dial-peer to match incoming calls, but in some cases, like yours, certain commands must be specified.
The "incoming called-number ." command is used to match the incoming calls to that dial-peer. The 'direct-inward-dial' command *should* eliminate that dialtone your caller hear. And these need to be applied to a pots dial-peer because the incoming call is coming in on a POTS pri.

Direct IP Dialing

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