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PSTN to CUCME

isaaco001
Level 3
Level 3

Dear Community,

I am able to make calls outside(pstn) using this dial peer below with gsm number.

dial-peer voice 4 pots
destination-pattern 9..........
incoming called-number .
port 0/3/3

I ,however, want calls to be received on a particular directory number  which should be my switchboard.I have seen this kind of configuration done for SKINNY with trunk 9 monitor port 0/3/3 command if this was the case.This configuration is done under ephone-dn and the command is available under ephone-dn command.SIP doesnt have this command under voice register dn.

How do i map outside calls to a particular directory number in SIP without DID number?Is this the correct dial-peer to receive outside calls?

Attached are the full configs.Thank you.

2 Accepted Solutions

Accepted Solutions

Suresh Hudda
VIP Alumni
VIP Alumni
Seems u r trying it on fxo port, if yes..

voice-port 0/3/3 cptone KE
Connection plar opx <ext number> description Safaricom caller-id enable

For incoming call through FXO line you can use connection plar command as below to point it to any specific extension.

Suresh

View solution in original post

Dear, I said it depends. You may need it to change CLI, lSDN plan, type etc if required. What yours service said to about outgoing calling.

Suresh

View solution in original post

22 Replies 22

Suresh Hudda
VIP Alumni
VIP Alumni
Seems u r trying it on fxo port, if yes..

voice-port 0/3/3 cptone KE
Connection plar opx <ext number> description Safaricom caller-id enable

For incoming call through FXO line you can use connection plar command as below to point it to any specific extension.

Suresh

Yes its an FXO port and the command connection plar opx worked.Now i can make calls from pstn to a particular extension.Thanks you.

However,you must have seen in the configs that i will be using E1 port later.can the command connection plar opx still be used with E1 port?Thank you once again.

Nope, you can't use this command on E1 pri. For that you need to apply translation rule/profile.

Suresh

Just to clarify,Will i use translation rule/profile for both incoming calls and outgoing calls or just incoming calls for pstn?.Thanks.

It depends, for incoming calls you need to make translation rule for called number to directly connect these calls to extensions. If you want to change CLI for outbound calls ( or want to change isdn plan and type; but this requires only when you need to change them in case of any issue) you need make translation rule for calling numbers, so this is not mandatory.

Suresh

Hi,i have started configuring E1 connection and i am not conversant with it.However,i followed some instructions and i managed to get this from show isdn status command:

ISDN Serial0/1/0:15 interface
dsl 0, interface ISDN Switchtype = primary-ni
Layer 1 Status:
ACTIVE
Layer 2 Status:
TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
Layer 3 Status:
0 Active Layer 3 Call(s)
Active dsl 0 CCBs = 0
The Free Channel Mask: 0xFFFF7FFF
Number of L2 Discards = 0, L2 Session ID = 2
Total Allocated ISDN CCBs = 0

The multiple_frame_established is a good sign,i think.This is how i configured the E1 controller:

E1 0/1/0 is up.
Applique type is Channelized E1 - balanced
No alarms detected.
alarm-trigger is not set
Version info FPGA Rev: 08121917, FPGA Type: PRK4
Framing is NO-CRC4, Line Code is HDB3, Clock Source is Line.
International Bit: 1, National Bits: 11111
Data in current interval (850 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
Data in Interval 1:
0 Line Code Violations, 0 Path Code Violations
1 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
1 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 413 Unavail Secs
Data in Interval 2:
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, 900 Unavail Secs
Data in Interval 3:
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, 197 Unavail Secs
Data in Interval 4:
4878 Line Code Violations, 11 Path Code Violations
0 Slip Secs, 0 Fr Loss Secs, 1 Line Err Secs, 0 Degraded Mins
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 556 Unavail Secs
Data in Interval 5:
25 Line Code Violations, 0 Path Code Violations
0 Slip Secs, 0 Fr Loss Secs, 1 Line Err Secs, 0 Degraded Mins
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 900 Unavail Secs
Total Data (last 5 15 minute intervals):
4903 Line Code Violations, 11 Path Code Violations,
1 Slip Secs, 0 Fr Loss Secs, 2 Line Err Secs, 0 Degraded Mins,
1 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 2966 Unavail Secs

i configured it with pri-group timeslots 1-31 

ISDN 0/1/0:15 Slot is 0, Subslot is 1, Sub-unit is 0, Port is 15
Type of VoicePort is ISDN-VOICE
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Description is not set
Noise Regeneration is enabled
Non Linear Processing is enabled
Non Linear Mute is disabled
Non Linear Threshold is -21 dB
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancellation NLP mute is disabled
Echo Cancellation NLP threshold is -21 dB
Echo Cancel Coverage is set to 128 ms
Echo Cancel worst case ERL is set to 6 dB
Playout-delay Mode is set to adaptive
Playout-delay Nominal is set to 60 ms
Playout-delay Maximum is set to 1000 ms
Playout-delay Minimum mode is set to default, value 40 ms
Playout-delay Fax is set to 300 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 15 s
Interdigit Time Out is set to 10 s
Call Disconnect Time Out is set to 60 s
Ringing Time Out is set to 180 s
Wait Release Time Out is set to 30 s
Companding Type is A-law
Region Tone is set for US
Station name None, Station number None
Translation profile (Incoming):
Translation profile (Outgoing):
Voice class called number pool:
lpcor (Incoming):
lpcor (Outgoing):


DS0 channel specific status info:
IN OUT
PORT CH SIG-TYPE OPER STATUS STATUS TIP RING
=============== == ============ ==== ====== ====== === ====
0/1/0:15 01 isdn-voice dorm none none
0/1/0:15 02 isdn-voice dorm none none
0/1/0:15 03 isdn-voice dorm none none
0/1/0:15 04 isdn-voice dorm none none
0/1/0:15 05 isdn-voice dorm none none
0/1/0:15 06 isdn-voice dorm none none
0/1/0:15 07 isdn-voice dorm none none
0/1/0:15 08 isdn-voice dorm none none
0/1/0:15 09 isdn-voice dorm none none
0/1/0:15 10 isdn-voice dorm none none
0/1/0:15 11 isdn-voice dorm none none
0/1/0:15 12 isdn-voice dorm none none
0/1/0:15 13 isdn-voice dorm none none
0/1/0:15 14 isdn-voice dorm none none
0/1/0:15 15 isdn-voice dorm none none
0/1/0:15 17 isdn-voice dorm none none
0/1/0:15 18 isdn-voice dorm none none
0/1/0:15 19 isdn-voice dorm none none
0/1/0:15 20 isdn-voice dorm none none
0/1/0:15 21 isdn-voice dorm none none
0/1/0:15 22 isdn-voice dorm none none
0/1/0:15 23 isdn-voice dorm none none
0/1/0:15 24 isdn-voice dorm none none
0/1/0:15 25 isdn-voice dorm none none
0/1/0:15 26 isdn-voice dorm none none
0/1/0:15 27 isdn-voice dorm none none
0/1/0:15 28 isdn-voice dorm none none
0/1/0:15 29 isdn-voice dorm none none
0/1/0:15 30 isdn-voice dorm none none
0/1/0:15 31 isdn-voice dorm none none

Please find full configs attached with dial-peer for E1,I expected outside calls to pstn at  least work without a translation profile.Thank you, please guide. 

dial-peer voice 7 pots
destination-pattern 2T
incoming called-number .
direct-inward-dial
port 0/1/0:15

Please attach debug isdn q931 && debug voip ccapi inout along with called number for a test outbound call. Suresh

I will attach debugs tomorrow,please have look at the provider settings attached(screen shots) if they may affect dial-peer operation.Thanks.

Please find attached debugs for isdn q931 and voip ccapi inout,i'll keep troubleshooting with given output codes,thanks.

You are sending 100 as calling number, can you check with service provider what they are supposed to get in calling number. And please try to change the isdn switch type to something else once and test.

Suresh

hi,the service provider calling number should be  07xxxxxx000 ,i have used a translation profile to change any internal extensions like 100  to this number i.e

rule 1 /.*/ /7xxxxxx000/  

However,i still cant make calls to or from pstn,i have tried changing the isdn-switch type but still no success.The isdn switch was set with isdn switch-type  primary-ni as show in the attached screenshots.I am still consulting with the provider,attached are the debugs.Thanks once again.

Okay, it is good to check with service provider that why they are sending DISCONNECT message.

Suresh

Hi,(update)incoming calls are working fine after i used a translation rule to convert 10 digit number to 3 digit extensions.However,outgoing calls still have an issue,am inquiring from the provider about the issue.I have tried using a translation rule for outgoing calls but still no success and as you stated in the earlier threads,i don;t really need a translation rule for outbound calls right?

Dear, I said it depends. You may need it to change CLI, lSDN plan, type etc if required. What yours service said to about outgoing calling.

Suresh