ESRST and translation-profile

[Thomas P]

Hello,

I have an ISR 2911 with the IOS 15.3(3)M5. This gateway is configured to allow SIP Phones to register in case of a CUCM failure using the eSRST mode.

Here is the configuration:

voice register global
 mode esrst
 system message "Mode SRST"
 max-dn 200
 max-pool 50
 timezone 2
!
voice register pool  1
 id network 10.X.X.X mask 255.255.255.0
 dtmf-relay rtp-nte sip-notify
 voice-class codec 1
!

My problem is that I cannot configure a translation-profile under the voice register pool : the command does not exist anymore.

Am I facing a bug or is there a new way to configure a translation profile for SIP Phones in SRST?

Thank you for your help.

Thomas.

[Terry Cheema]

I havent tried under voice register pool for a while so not sure why this is not available, but you can apply translation-profile under your appropriate dial peers as well.

-Terry

[Thomas P]

Thank you for your answer.

The goal of my translation-profile is to allow people to dial short numbers to call their colleagues internally (for instance, dial 456 to call 457896532). So, I don't think I can apply this kind of translation-profile under a dial-peer?

Thomas.

[lbjr]

Have you ever gotten an answer to this, it probably can be accomplished with dial peers  and your translation profile plus a  higher Preference no. in SRST. But I am curious if the command was depreciated or something. I also don't understand why mode choice is ESRST and no longer  SRST

[Thomas P]

Hello,

The solution for me was to go back to srst mode by using the command "no mode esrst" under the voice register global.

I am interested to know how to accomplish that with dial peers.

[lbjr]

I would just do

voice translation rule 99

rule 1 /456/ /457896532/

Voice translation-profile shortdial

translate called 99

dial-peer voice XX voip

destination-pattern 456

translation-profile outgoing shortdial

session target ipv4:x.x.x.x (SRST Router)

dtmf-relay h245 alpha

codec g711u

no vad

pref x

If you dial 456 while in SRST in wont find it as an ephone or in the SIP address pool, so therefore it should choose the above Dial peer.

[Thomas P]

Yes, you're wright. It's also possible to do this with dial-peer.

Thanks for your answer.

[orlandoguy82]

I found making 4-digit short dial for +e164 DNs in esrst mode difficult to figure out.

One cannot add a translation-profile to the esrst voice register pool like with "mode srst" in fact one cannot choose "mode srst" at all, only esrst or cme. I used the following method to expand my 5[78].. range out to the full +e164 the phones have. These numbers can be seen as the created dial-peers using the "show sip-ua status registrar" as shown below:

RT01#show sip-ua status registrar
Line          destination      expires(sec)  contact
transport     call-id
              peer
============================================================
+12224445759  10.64.150.36     283           10.64.150.36
UDP           0041d2f8-0f92018d-20a58720-0994d939@10.64.150.36    
              40024

voice translation-rule 1

Rule 1 /^\(5[78]..\)$/ /+1222444\1/

!

!

voice translation-profile srst-expand-e164

translate called 1

!

dial-peer voice 10 voip

session protocol sipv2

destination-pattern 5[78]..$

translation-profile outgoing srst-expand-e164

session target ipv4:10.64.31.251     <<<<This is the router, the same IP SIP is bound to

dtmf-relay rtp-nte sip-kpml

codec g711u

no vad

Calls from SIP phones to 5700-5800 will expand out to +12224445000 and catch the dial-peers of the devices registered in esrst mode.

I hope this helps others.

[orlandoguy82]

I found making 4-digit short dial for +e164 DNs in esrst mode difficult to figure out.

One cannot add a translation-profile to the esrst voice register pool like with "mode srst" in fact one cannot choose "mode srst" at all, only esrst or cme. I used the following method to expand my 5[78].. range out to the full +e164 the phones have. These numbers can be seen as the created dial-peers using the "show sip-ua status registrar" as shown below:

RT01#show sip-ua status registrar
Line          destination      expires(sec)  contact
transport     call-id
              peer
============================================================
+12224445759  10.64.150.36     283           10.64.150.36
UDP           0041d2f8-0f92018d-20a58720-0994d939@10.64.150.36    
              40024

voice translation-rule 1

Rule 1 /^\(5[78]..\)$/ /+1222444\1/

!

!

voice translation-profile srst-expand-e164

translate called 1

!

dial-peer voice 10 voip

session protocol sipv2

destination-pattern 5[78]..$

translation-profile outgoing srst-expand-e164

session target ipv4:10.64.31.251     <<<<This is the router, the same IP SIP is bound to

dtmf-relay rtp-nte sip-kpml

codec g711u

no vad

Calls from SIP phones to 5700-5800 will expand out to +12224445000 and catch the dial-peers of the devices registered in esrst mode.

I hope this helps others.

[jwani]

You will first need to configure the alias command within the voice register pool in order to be able to use the translation-profile command.

[jeanm02]

I recently had the same issue on 8200 series... below is how I was able to resolve it.

voice register global #

default mode

no allow-hash-in-dn

system message SRST Active

max-dn XXX

max-pool XXX