Can't get inbound or outbound calls coming in but POTS line is working?

So I've been working on my CUCM lab and trying to make calls work outbound and receive from and to the PSTN but they are not working:
!
voice-port 1/1/0
connection plar opx 100
station-id number 100
!
voice-port 1/1/1
!
!
!
!
!
dial-peer voice 102 pots
destination-pattern 9[^1].........
port 1/1/0
!
dial-peer voice 103 pots
destination-pattern 911
port 1/1/0
forward-digits all
!
dial-peer voice 104 pots
destination-pattern 9678.......
port 1/1/0
!
dial-peer voice 100 pots
destination-pattern 9770.......
port 1/1/0
!
dial-peer voice 105 pots
destination-pattern 9404.......
port 1/1/0
!
dial-peer voice 106 pots
destination-pattern 91..........
port 1/1/0
!
dial-peer voice 107 pots
destination-pattern 9T
port 1/1/0
!
dial-peer voice 1 voip
destination-pattern 1..
voice-class h323 1
session target ipv4:10.0.1.1
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
I ran the show active voice calls command and got this when I was trying to make calls:
Telephony call-legs: 1
SIP call-legs: 0
H323 call-legs: 1
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Media call-legs: 0
Total call-legs: 2
GENERIC:
SetupTime=897479720 ms
Index=1
PeerAddress=100
PeerSubAddress=
PeerId=1
PeerIfIndex=16
LogicalIfIndex=0
ConnectTime=897482640 ms
CallDuration=00:00:21 sec
CallState=4
CallOrigin=2
ChargedUnits=0
InfoType=speech
TransmitPackets=1098
TransmitBytes=175680
ReceivePackets=1097
ReceiveBytes=175520
VOIP:
ConnectionId[0x800B4DBF 0x3C2531B3 0x21002101 0xA01010C]
IncomingConnectionId[0x800B4DBF 0x3C2531B3 0x21002101 0xA01010C]
CallID=72
RemoteIPAddress=10.0.1.1
RemoteUDPPort=17292
RemoteSignallingIPAddress=10.0.1.1
RemoteSignallingPort=52490
RemoteMediaIPAddress=10.1.1.12
RemoteMediaPort=17292
RoundTripDelay=0 ms
SelectedQoS=best-effort
tx_DtmfRelay=h245-alphanumeric
FastConnect=FALSE
Separate H245 Connection=TRUE
H245 Tunneling=FALSE
SessionProtocol=cisco
ProtocolCallId=
SessionTarget=
OnTimeRvPlayout=18100
GapFillWithSilence=0 ms
GapFillWithPrediction=0 ms
GapFillWithInterpolation=0 ms
GapFillWithRedundancy=0 ms
HiWaterPlayoutDelay=65 ms
LoWaterPlayoutDelay=55 ms
TxPakNumber=909
TxSignalPak=0
TxComfortNoisePak=0
TxDuration=18190
TxVoiceDuration=18190
RxPakNumber=908
RxSignalPak=0
RxComfortNoisePak=0
RxDuration=18150
RxVoiceDuration=18100
RxOutOfSeq=0
RxLatePak=0
RxEarlyPak=0
RxBadProtocol=0
PlayDelayCurrent=55
PlayDelayMin=55
PlayDelayMax=65
PlayDelayClockOffset=-124553562
PlayDelayJitter=0 ms
PlayErrPredictive=0
PlayErrInterpolative=0
PlayErrSilence=0
PlayErrBufferOverFlow=10
PlayErrRetroactive=0
PlayErrTalkspurt=0
OutSignalLevel=-41
InSignalLevel=-42
LevelTxPowerMean=0
LevelRxPowerMean=0
LevelBgNoise=0
ERLLevel=9
ACOMLevel=9
ErrRxDrop=0
ErrTxDrop=0
ErrTxControl=0
ErrRxControl=0
ReceiveDelay=55 ms
LostPackets=0
EarlyPackets=0
LatePackets=0
SRTP = off
TextRelay = off
VAD = disabled
CoderTypeRate=g711ulaw
CodecBytes=160
Media Setting=flow-through
CallerName=
CallerIDBlocked=False
OriginalCallingNumber=100
OriginalCallingOctet=0x0
OriginalCalledNumber=97705707990
OriginalCalledOctet=0xA8
OriginalRedirectCalledNumber=
OriginalRedirectCalledOctet=0xFF
TranslatedCallingNumber=100
TranslatedCallingOctet=0x0
TranslatedCalledNumber=97705707990
TranslatedCalledOctet=0xA8
TranslatedRedirectCalledNumber=
TranslatedRedirectCalledOctet=0xFF
GwReceivedCalledNumber=97705707990
GwReceivedCalledOctet3=0xA8
GwReceivedCallingNumber=100
GwReceivedCallingOctet3=0x0
GwReceivedCallingOctet3a=0x81
MediaInactiveDetected=no
MediaInactiveTimestamp=
MediaControlReceived=
LongDurationCallDetected=no
LongDurCallTimestamp=
LongDurcallDuration=
Username=
Then on my dial peers:
sh dial-peer voice summary
dial-peer hunt 0
AD PRE PASS OUT
TAG TYPE MIN OPER PREFIX DEST-PATTERN FER THRU SESS-TARGET STAT PORT
102 pots up up 9[^1]......... 0 up 1/1/0
103 pots up up 911 0 up 1/1/0
104 pots up up 9678....... 0 up 1/1/0
100 pots up up 9770....... 0 up 1/1/0
105 pots up up 9404....... 0 up 1/1/0
106 pots up up 91.......... 0 up 1/1/0
107 pots up up 9T 0 up 1/1/0
1 voip up up 1.. 0 syst ipv4:10.0.1.1
UITS-2621XM#
I don't understand why I can't make outbound calls and receive them from the outside. Can someone please help me understand why? Would translation rules be a cause of this? I don't know how to configure them if they are, and i don't understand them much. I see that my gateway is getting the incoming calls now but my phones aren't ringing or when I dial out the are ringing out, BUT I get this distorted noise. Any ideas? I see that my gateway is getting the incoming calls now but my phones aren't ringing or when I dial out the are ringing out, BUT I get this distorted noise. Any ideas?
!
voice-port 1/1/0
connection plar opx 100
station-id number 100
!
voice-port 1/1/1
!
!
!
!
!
dial-peer voice 102 pots
destination-pattern 9[^1].........
port 1/1/0
!
dial-peer voice 103 pots
destination-pattern 911
port 1/1/0
forward-digits all
!
dial-peer voice 104 pots
destination-pattern 9678.......
port 1/1/0
!
dial-peer voice 100 pots
destination-pattern 9770.......
port 1/1/0
!
dial-peer voice 105 pots
destination-pattern 9404.......
port 1/1/0
!
dial-peer voice 106 pots
destination-pattern 91..........
port 1/1/0
!
dial-peer voice 107 pots
destination-pattern 9T
port 1/1/0
!
dial-peer voice 1 voip
destination-pattern 1..
voice-class h323 1
session target ipv4:10.0.1.1
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
I ran the show active voice calls command and got this when I was trying to make calls:
Telephony call-legs: 1
SIP call-legs: 0
H323 call-legs: 1
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Media call-legs: 0
Total call-legs: 2
GENERIC:
SetupTime=897479720 ms
Index=1
PeerAddress=100
PeerSubAddress=
PeerId=1
PeerIfIndex=16
LogicalIfIndex=0
ConnectTime=897482640 ms
CallDuration=00:00:21 sec
CallState=4
CallOrigin=2
ChargedUnits=0
InfoType=speech
TransmitPackets=1098
TransmitBytes=175680
ReceivePackets=1097
ReceiveBytes=175520
VOIP:
ConnectionId[0x800B4DBF 0x3C2531B3 0x21002101 0xA01010C]
IncomingConnectionId[0x800B4DBF 0x3C2531B3 0x21002101 0xA01010C]
CallID=72
RemoteIPAddress=10.0.1.1
RemoteUDPPort=17292
RemoteSignallingIPAddress=10.0.1.1
RemoteSignallingPort=52490
RemoteMediaIPAddress=10.1.1.12
RemoteMediaPort=17292
RoundTripDelay=0 ms
SelectedQoS=best-effort
tx_DtmfRelay=h245-alphanumeric
FastConnect=FALSE
Separate H245 Connection=TRUE
H245 Tunneling=FALSE
SessionProtocol=cisco
ProtocolCallId=
SessionTarget=
OnTimeRvPlayout=18100
GapFillWithSilence=0 ms
GapFillWithPrediction=0 ms
GapFillWithInterpolation=0 ms
GapFillWithRedundancy=0 ms
HiWaterPlayoutDelay=65 ms
LoWaterPlayoutDelay=55 ms
TxPakNumber=909
TxSignalPak=0
TxComfortNoisePak=0
TxDuration=18190
TxVoiceDuration=18190
RxPakNumber=908
RxSignalPak=0
RxComfortNoisePak=0
RxDuration=18150
RxVoiceDuration=18100
RxOutOfSeq=0
RxLatePak=0
RxEarlyPak=0
RxBadProtocol=0
PlayDelayCurrent=55
PlayDelayMin=55
PlayDelayMax=65
PlayDelayClockOffset=-124553562
PlayDelayJitter=0 ms
PlayErrPredictive=0
PlayErrInterpolative=0
PlayErrSilence=0
PlayErrBufferOverFlow=10
PlayErrRetroactive=0
PlayErrTalkspurt=0
OutSignalLevel=-41
InSignalLevel=-42
LevelTxPowerMean=0
LevelRxPowerMean=0
LevelBgNoise=0
ERLLevel=9
ACOMLevel=9
ErrRxDrop=0
ErrTxDrop=0
ErrTxControl=0
ErrRxControl=0
ReceiveDelay=55 ms
LostPackets=0
EarlyPackets=0
LatePackets=0
SRTP = off
TextRelay = off
VAD = disabled
CoderTypeRate=g711ulaw
CodecBytes=160
Media Setting=flow-through
CallerName=
CallerIDBlocked=False
OriginalCallingNumber=100
OriginalCallingOctet=0x0
OriginalCalledNumber=97705707990
OriginalCalledOctet=0xA8
OriginalRedirectCalledNumber=
OriginalRedirectCalledOctet=0xFF
TranslatedCallingNumber=100
TranslatedCallingOctet=0x0
TranslatedCalledNumber=97705707990
TranslatedCalledOctet=0xA8
TranslatedRedirectCalledNumber=
TranslatedRedirectCalledOctet=0xFF
GwReceivedCalledNumber=97705707990
GwReceivedCalledOctet3=0xA8
GwReceivedCallingNumber=100
GwReceivedCallingOctet3=0x0
GwReceivedCallingOctet3a=0x81
MediaInactiveDetected=no
MediaInactiveTimestamp=
MediaControlReceived=
LongDurationCallDetected=no
LongDurCallTimestamp=
LongDurcallDuration=
Username=
Then on my dial peers:
sh dial-peer voice summary
dial-peer hunt 0
AD PRE PASS OUT
TAG TYPE MIN OPER PREFIX DEST-PATTERN FER THRU SESS-TARGET STAT PORT
102 pots up up 9[^1]......... 0 up 1/1/0
103 pots up up 911 0 up 1/1/0
104 pots up up 9678....... 0 up 1/1/0
100 pots up up 9770....... 0 up 1/1/0
105 pots up up 9404....... 0 up 1/1/0
106 pots up up 91.......... 0 up 1/1/0
107 pots up up 9T 0 up 1/1/0
1 voip up up 1.. 0 syst ipv4:10.0.1.1
UITS-2621XM#
I don't understand why I can't make outbound calls and receive them from the outside. Can someone please help me understand why? Would translation rules be a cause of this? I don't know how to configure them if they are, and i don't understand them much. I see that my gateway is getting the incoming calls now but my phones aren't ringing or when I dial out the are ringing out, BUT I get this distorted noise. Any ideas? I see that my gateway is getting the incoming calls now but my phones aren't ringing or when I dial out the are ringing out, BUT I get this distorted noise. Any ideas?
Comments
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pitviper Member Posts: 1,376 ■■■■■■■□□□
Are you sending the call to a UCM server, or CME router? If UCM, how did you setup the H323 GW? Do you have a phone that's extension "100"?
Also check the digit-strip characteristics for POTS dial-peers. Anything matching 100, 104, or 105 is only sending 7 digits to the PSTN – Is this what you want?CCNP:Collaboration, CCNP:R&S, CCNA:S, CCNA:V, CCNA, CCENT -
khayes Member Posts: 16 ■□□□□□□□□□
Are you sending the call to a UCM server, or CME router? If UCM, how did you setup the H323 GW? Do you have a phone that's extension "100"?
Also check the digit-strip characteristics for POTS dial-peers. Anything matching 100, 104, or 105 is only sending 7 digits to the PSTN – Is this what you want?
The CUCM server is a VMware server, and the H323 gateway is running on my Cisco 2621XM router. My location requires 10 digit dialing for local calling. I need 10 digits to work inbound and Outbound.
I have my own extention, my wife's and the CIPCC extentions.
My Gateway and my CUCM server are talking. I know the calls come in to the gateway there's activity BUT none of the phones are ringing. -
pitviper Member Posts: 1,376 ■■■■■■■□□□
The CUCM server is a VMware server, and the H323 gateway is running on my Cisco 2621XM router. My location requires 10 digit dialing for local calling. I need 10 digits to work inbound and Outbound.
I have my own extention, my wife's and the CIPCC extentions.
My Gateway and my CUCM server are talking. I know the calls come in to the gateway there's activity BUT none of the phones are ringing.
OK, for inbound on a POTS line you’re not going to get dialed number info – that’s why you send it directly to “somewhere” via the FXO config with the PLAR command. The call is hitting the gateway and forwarded to extension 100 – is that a dial-able extension on the UCM box? If not, you either have to send the call to a valid extension (or hunt pilot, and so on) or you’ll need to translate it (either on the gateway or in UCM).
For outbound, POTS dial-peers will by default strip off explicitly defined digits – If you make a local call to area codes 678, 770, or 404 only 7 digits are being sent. With your long distance dial-peer, the 1 is being stripped off. Need to add “forward-digits [number]”
Have you studied the CVOICE material? My advice would be to focus on CVOICE using CME first, then apply the knowledge to UCM later down the road. Call Manager is a monster – you really need a solid foundation before you dive in.CCNP:Collaboration, CCNP:R&S, CCNA:S, CCNA:V, CCNA, CCENT -
khayes Member Posts: 16 ■□□□□□□□□□
OK, for inbound on a POTS line you’re not going to get dialed number info – that’s why you send it directly to “somewhere” via the FXO config with the PLAR command. The call is hitting the gateway and forwarded to extension 100 – is that a dial-able extension on the UCM box? If not, you either have to send the call to a valid extension (or hunt pilot, and so on) or you’ll need to translate it (either on the gateway or in UCM).
For outbound, POTS dial-peers will by default strip off explicitly defined digits – If you make a local call to area codes 678, 770, or 404 only 7 digits are being sent. With your long distance dial-peer, the 1 is being stripped off. Need to add “forward-digits [number]”
Have you studied the CVOICE material? My advice would be to focus on CVOICE using CME first, then apply the knowledge to UCM later down the road. Call Manager is a monster – you really need a solid foundation before you dive in.
I got outbound working not inbound yet. I do plan on learning CME after I get the CUCM running. -
khayes Member Posts: 16 ■□□□□□□□□□
OK, for inbound on a POTS line you’re not going to get dialed number info – that’s why you send it directly to “somewhere” via the FXO config with the PLAR command. The call is hitting the gateway and forwarded to extension 100 – is that a dial-able extension on the UCM box? If not, you either have to send the call to a valid extension (or hunt pilot, and so on) or you’ll need to translate it (either on the gateway or in UCM).
For outbound, POTS dial-peers will by default strip off explicitly defined digits – If you make a local call to area codes 678, 770, or 404 only 7 digits are being sent. With your long distance dial-peer, the 1 is being stripped off. Need to add “forward-digits [number]”
Have you studied the CVOICE material? My advice would be to focus on CVOICE using CME first, then apply the knowledge to UCM later down the road. Call Manager is a monster – you really need a solid foundation before you dive in.
I've been studying the CVOICE material off and on. First of the year I plan on starting full-blown, but in the meantime I prefer to play with the lab first and learn as I go. It helps me understand the CVOICE stuff better. -
Luckycharms Member Posts: 267
Always remember that a Call has at least two call legs.. (inbound and outbound) 9/10 people will use a single dial-peer for both...but to fix your inbound dial problem read up on " Direct Inward Dial" ...Understanding Direct-Inward-Dial (DID) on IOS Voice Digital (T1/E1) Interfaces - Cisco SystemsThe quality of a book is never equated to the number of words it contains. -- And neither should be a man by the number of certifications or degree's he has earned.