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CallerID SRST inbound and outbound

latintrpt
Level 1
Level 1

Hello,

I got SRST working on 2921 we have for a WAN location.  We have two PRI's at the location and all DID's are held at that site.

Phone calls in SRST are working fine inbound and outbound.

My problem is that when I call a Cisco IP phone that is in SRST from my cell phone it, on the Cisco IP phone it shows up as: 773XXXXXXX.  Is there anyway to prepend 91 so that it shows up as 91773XXXXXXX ?

My other problem is that when I call my cell phone from a Cisco IP phone that is in SRST, the number on my cell phone shows up as: 4-4503, the 5 digit number.  Is there any way to get it to show the whole number: 53072444503.  The telco by is sending us 5 digits by the way.

Any help would be appreciated.

Thanks

1 Accepted Solution

Accepted Solutions

This is the essential parts of your config for just the number manipulation.

voice translation-rule 1

rule 7 /\(1..........\)/ /9\1/

rule 8 /\(..........\)/ /91\1/

!

voice translation-rule 2

rule 1 /^\(4....\)$/ /63072\1/

!

voice translation-rule 10

rule 1 /^91\(..........\)$/ /\1/

rule 2 /^91\(.*\)/ /\1/

!

!

voice translation-profile CALLID_Outgoing_SRST

translate calling 2

!

voice translation-profile INBOUND_PSTN

translate calling 1

!

voice translation-profile REMOVE-PREFIX

translate calling 10

voice-port 0/0/0:23

translation-profile outgoing CALLID_Outgoing_SRST

bearer-cap Speech

voice-port 0/0/1:23

translation-profile outgoing CALLID_Outgoing_SRST

bearer-cap Speech

dial-peer voice 99 pots

description Calls from PRI into system

translation-profile incoming INBOUND_PSTN

incoming called-number .

direct-inward-dial

dial-peer voice 50 voip

description Inbound calls send to CCM

translation-profile outgoing 10

destination-pattern 4....

progress_ind setup enable 3

session target ipv4:10.74.213.235

voice-class codec 100

voice-class h323 1

dtmf-relay h245-alphanumeric

ip qos dscp cs5 media

ip qos dscp cs3 signaling

no vad

dial-peer voice 51 voip

description Inbound calls send to CCM

translation-profile outgoing 10

destination-pattern 4....

progress_ind setup enable 3

session target ipv4:10.74.213.236

voice-class codec 100

voice-class h323 1

dtmf-relay h245-alphanumeric

ip qos dscp cs5 media

ip qos dscp cs3 signaling

no vad

dial-peer voice 52 voip

description Inbound calls send to CCM

translation-profile outgoing 10

destination-pattern 4....

progress_ind setup enable 3

session target ipv4:10.74.213.238

voice-class codec 100

voice-class h323 1

dtmf-relay h245-alphanumeric

ip qos dscp cs5 media

ip qos dscp cs3 signaling

no vad

ephone-dn-template  1

translation-profile incoming INBOUND_PSTN

translation-profile outgoing CALLID_Outgoing_SRST

I've highlighted the parts that are of interest with different colors. The blue highlight is the name on your new voice translation profile, the red part is the error you made in your config and the orange part is for config that you probably want to remove, since it’s redundant.

The change you need to do to fix your misconfiguration is the below part.

dial-peer voice 50 voip

translation-profile outgoing REMOVE-PREFIX

dial-peer voice 51 voip

translation-profile outgoing REMOVE-PREFIX

dial-peer voice 52 voip

translation-profile outgoing REMOVE-PREFIX

Please remember to rate helpful responses and identify helpful or correct answers.



Response Signature


View solution in original post

29 Replies 29

Chris Deren
Hall of Fame
Hall of Fame

Yes, use translation-profiles with translation-rules and apply it to call-manager-fallback, similar question has been asked here many times so a search should reveal many good threads.

HTH,

Chris

we don't use call-manager-fallback.  We are using CUCME on the 2921.

Thanks

The you do it under telephony-service.

HTH,

Chris

there is no option to add a translation profile to the telephony-service.

Would have I have to add it to the ephone-dn?

Thank You

Use voice translation rules tied to either incoming/outgoing dial peers or direct on the voice port.

Please remember to rate all useful posts.

Sent from Cisco Technical Support iPhone App



Response Signature


If you're using CME in SRST mode you would need to use an ephone-dn template and apply the translation profile there. Once that exists you then go under telephony-service to define that template for use by SRST phones.

ephone-dn-template  1

translation-profile incoming DN_e164_Incoming

translation-profile outgoing DN_e164_Outgoing

huntstop channel

telephony-service

srst mode auto-provision none

srst ephone template 1

srst dn template 1

srst dn line-mode dual

max-ephones 40

max-dn 100

Please remember to rate helpful responses and identify helpful or correct answers.

I'm still having issues

Here is what I have:

------------------------------------------------------------------------------------------------------------------------------------------------------------------------------

voice translation-rule 1

rule 1 /\(^[2-9]&$\)/ /9\1/

rule 2 /\(^[2-9]&\)/ /91\1/

rule 3 /\(011.*\)/ /9\1/

!

voice translation-rule 2

rule 1 /^\(4....\)$/ /630724\1/

!

!

voice translation-profile CALLID_Outgoing_SRST

translate calling 2

!

voice translation-profile PSTN_Incoming_SRST_PREPEND_91

translate calling 1

voice-port 0/0/0:23

bearer-cap Speech

!

voice-port 0/1/0

!

voice-port 0/1/1

!

voice-port 0/1/2

!

voice-port 0/1/3

!

voice-port 0/0/1:23

bearer-cap Speech

dial-peer voice 50 voip

description Inbound calls send to CCM

destination-pattern 4....

progress_ind setup enable 3

session target ipv4:10.74.213.235

voice-class codec 100

voice-class h323 1

dtmf-relay h245-alphanumeric

ip qos dscp cs5 media

ip qos dscp cs3 signaling

no vad

!

dial-peer voice 51 voip

description Inbound calls send to CCM

preference 1

destination-pattern 4....

progress_ind setup enable 3

session target ipv4:10.74.13.236

voice-class codec 100

voice-class h323 1

dtmf-relay h245-alphanumeric

ip qos dscp cs5 media

ip qos dscp cs3 signaling

no vad

!

dial-peer voice 52 voip

description Inbound calls send to CCM

preference 2

destination-pattern 4....

progress_ind setup enable 3

session target ipv4:10.74.13.228

voice-class codec 100

voice-class h323 1

dtmf-relay h245-alphanumeric

ip qos dscp cs5 media

ip qos dscp cs3 signaling

no vad

!

dial-peer voice 3 pots

trunkgroup PRI

destination-pattern 91[2-9]..[2-9]......

progress_ind alert enable 8

progress_ind progress enable 8

progress_ind connect enable 8

forward-digits 11

!

dial-peer voice 911 pots

trunkgroup PRI

description Emergency Calls

destination-pattern 911

progress_ind alert enable 8

progress_ind progress enable 8

progress_ind connect enable 8

forward-digits 3

!

dial-peer voice 9911 pots

trunkgroup PRI

description Emergency Calls

destination-pattern 9911

progress_ind alert enable 8

progress_ind progress enable 8

progress_ind connect enable 8

forward-digits 3

!

dial-peer voice 99 pots

description Calls from PRI into system

incoming called-number .

direct-inward-dial

!

dial-peer voice 4 pots

trunkgroup PRI

destination-pattern 311

progress_ind alert enable 8

progress_ind progress enable 8

progress_ind connect enable 8

forward-digits 3

!

dial-peer voice 5 pots

trunkgroup PRI

destination-pattern 411

progress_ind alert enable 8

progress_ind progress enable 8

progress_ind connect enable 8

forward-digits 3

!

telephony-service

srst mode auto-provision all

srst ephone template 5

srst ephone description srst fallback auto-provision phone : Nov 14 2012 20:53:

26

srst dn template 1

srst dn line-mode dual

max-ephones 30

max-dn 60

ip source-address 172.18.0.140 port 2000

max-redirect 20

system message "You are in Survivablity MODE"

keepalive 10

max-conferences 8 gain -6

moh "welcome.au"

transfer-system full-consult

create cnf-files version-stamp 7960 Nov 30 2012 00:42:23

!

!

ephone-dn-template  1

translation-profile incoming PSTN_Incoming_SRST_PREPEND_91

translation-profile outgoing CALLID_Outgoing_SRST

huntstop channel

!

!

ephone-template  5

softkeys idle  Newcall Cfwdall Redial Pickup

softkeys seized  Endcall Cfwdall Pickup

softkeys alerting  Endcall

softkeys connected  Endcall Hold Park Trnsfer Confrn ConfList

type 7962

!

ephone-dn  6  dual-line

number 45054

label 45054

description 6307248700

name 6307248700

ephone-dn-template 1

!

ephone  6

description srst fallback auto-provision phone : Nov 14 2012 20:53:26

mac-address 34BD.C82D.9E7D

ephone-template 5

type 7962

button  1:6

I excluded the other ephone and ephone-dn's except for 6.

Anything that would still cause this to fail?

Thanks

I believe I have this working now:

-----------------------------------------------------------------------------------------------------------------------------------------------------------------------------

voice translation-rule 1

rule 7 /\(1..........\)/ /9\1/

rule 8 /\(..........\)/ /91\1/

!

voice translation-rule 2

rule 1 /^\(4....\)$/ /63072\1/

!

!

voice translation-profile CALLID_Outgoing_SRST

translate calling 2

!

voice translation-profile INBOUND_PSTN

translate calling 1

voice-port 0/0/0:23

translation-profile outgoing CALLID_Outgoing_SRST

bearer-cap Speech

voice-port 0/0/1:23

translation-profile outgoing CALLID_Outgoing_SRST

bearer-cap Speech

dial-peer voice 99 pots

description Calls from PRI into system

translation-profile incoming INBOUND_PSTN

incoming called-number .

direct-inward-dial

telephony-service

srst mode auto-provision all

srst ephone template 5

srst ephone description srst fallback auto-provision phone : Nov 14 2012 20:53:

26

srst dn template 1

srst dn line-mode dual

max-ephones 30

max-dn 60

ip source-address 172.18.0.140 port 2000

max-redirect 20

system message "You are in Survivablity MODE"

keepalive 10

max-conferences 8 gain -6

moh "welcome.au"

transfer-system full-consult

create cnf-files version-stamp 7960 Nov 30 2012 00:42:23

!

!

ephone-dn-template  1

translation-profile incoming INBOUND_PSTN

translation-profile outgoing CALLID_Outgoing_SRST

huntstop channel

!

!

ephone-template  5

softkeys idle  Newcall Cfwdall Redial Pickup

softkeys seized  Endcall Cfwdall Pickup

softkeys alerting  Endcall

softkeys connected  Endcall Hold Park Trnsfer Confrn ConfList

type 7962

!

ephone-dn  6  dual-line

number 45054

label 45054

description 6307248700

name 6307248700

ephone-dn-template 1

!

ephone  6

description srst fallback auto-provision phone : Nov 14 2012 20:53:26

mac-address 34BD.C82D.9E7D

ephone-template 5

type 7962

button  1:6

-----------------------------------------------------------------------------------------------------------------------------------------------------------------------------

Anybody else have another way they would've done it?

Thanks

Hi,

There is nothing wrong using voice translation rules on your dial-peers to do this as you have just done. However, there are a few things to consider

1. Using xation rules on your dial-peers and voice ports may impact in how CLI is presented during normal working operations, if you are using h323 gateway. If yo ar eusing gcp gateway then you should be fine

2. There is no need to apply xlation rule on dial-peer, voice p orts and ephone-dn template...You can either do this on ephone-dn template or on dial-peers or on voice  ports..But it is not good to configure your xlation rules every where, this can create mor troubles for you especially when troubleshooting.

Please rate all useful posts

"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

Please rate all useful posts

Hi aokanlawon,

you are correct about number 1 as this is exactly whta I'm running to right now during normal operations (non-srst mode). It's screwing my caller ID, calls are coming in 919773XXXXXXX. So how would I correct this issue?  I  tried just applying it to the ephone-template and it's not prepending  the 91 to calls coming from the PSTN to the Cisco phone in SRST.

I'm using the gateway as H.323 btw.

Hi,

OK. Try and remove the xlation rules from the dial-peer and the voice port. Then send the output of the ff:

show call-manager-fallback dial-peer

Please rate all useful posts

"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

Please rate all useful posts

I'm not using call-manager-fallback

Sorry about that.. Please use

show telephony-service dial-peer

Please rate all useful posts

"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

Please rate all useful posts
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