H.323-trunk -> works, but no function if an extension is forwarded

[DetNit1101]

Hello community,

i´ve set up a trunk between two UC500 with H.323.

Everything works fine, extensions are translated with correct node-numbers and so on,

calls are possible and missed calls can be answered correctly.

But if anyone calls over the trunk an extension which is forwarded (no matter: all, busy or noan to VoiceMail or to an other extension)

it doesn´t work; the A-side is disconnected immediately or after the noan-timer.

But if I call an free extension on side B and this extension makes a call transfer to an forwarded extension,

then it will work and I get the VoiceMail-Announcement on the side A.

The only hint with deb voice ccapi error

000147: Mar 11 11:48:45.235: //-1/xxxxxxxxxxxx/CCAPI/cc_validate_mlpp_info:
   Unsupported MLPP Service Domain Network 0

From internal extensions or PSTN does it work with call forwarding.

NODE 301
!
!
dial-peer voice 71 voip

description : SIP TOKYO
translation-profile incoming ankommend_von_TOKYO
translation-profile outgoing abgehend_nach_TOKYO
destination-pattern 501..
session target ipv4:11.2.10.198
incoming called-number 301..
dtmf-relay h245-alphanumeric
codec g711ulaw
supplementary-service h450.12
no supplementary-service h225-notify cid-update
!
!
<...>
!
!
telephony-service
...
transfer-system full-consult
transfer-pattern .T
...
!

#######################################################

NODE 501
!
!
dial-peer voice 90 voip
description : SIP MOSKAU
translation-profile incoming ankommend_von_MOSKAU
translation-profile outgoing abgehend_nach_MOSKAU
destination-pattern 301..
session target ipv4:11.2.11.196
incoming called-number 501..
codec g711ulaw
dtmf-relay h245-alphanumeric
supplementary-service h450.12
no supplementary-service h225-notify cid-update
!
!
<...>
!
!
telephony-service
...
transfer-system full-consult
transfer-pattern .T
...
!

also implemented on both systems:

!
voice service voip
ip address trusted list
  ipv4 11.2.11.192 255.255.255.192  / ipv4 11.2.10.192 255.255.255.192
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
supplementary-service ringback h225-info

Kind regards,

DetNit

[DetNit1101]

Hi,

according to the way that I received an info on node 301

that the original call was redirected to extension 55 (CUE)

on node 501, I create the following config on node 301:

!
!
dial-peer voice 55 voip
translation-profile outgoing redirect_to_CUE_501
destination-pattern 55
session target ipv4:10.212.10.198
codec g711ulaw
supplementary-service h450.12
!

!
voice translation-profile redirect_to_CUE_501
translate calling 71
translate called 73
!
!
voice translation-rule 71
rule 1 /^\(..\)/ /301\1/
!
!
voice translation-rule 73
rule 1 /^\(..\)/ /501\1/
!

And it works, I got the right mailbox 33 on node 501!

Or is this just a lousy work-around?

Kind regards,

DetNit

[David Trad]

Hi DetNit,

OK best thing for me to do is give you an example of the WAN-2-WAN Configuration I have used, this may or may not help you but it could set you on the right path.... NOTE: Transcoding is required for this especially when Voice Mail comes into play, so you will need to get that working correctly.

After the configuratione example i will answer your above questions

Site A:

voice call send-alert
voice rtp send-recv
!
voice service voip 
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 supplementary-service h450.12
 sip
  no update-callerid
!
!
!
voice class codec 1
 codec preference 1 g711alaw
 codec preference 2 g711ulaw
 codec preference 3 g729r8
voice class cause-code 1
 no-circuit
voice translation-rule 2002
 rule 1 /^6/ //
voice translation-profile XFER_TO_VM_PROFILE
 translate redirect-called 2002
This is the Voice Mail Pilot
dial-peer voice 2000 voip
 destination-pattern 699
 session protocol sipv2
 session target ipv4:10.1.10.1
 dtmf-relay sip-notify
 codec g711ulaw
This is the Dial-Peers for Calling Site-2, you will need to modify this accordingly
NOTE: The destination pattern is the DN number for the remote site
dial-peer voice 2010 voip
 description Incoming Call
 incoming called-number .%
!
dial-peer voice 2011 voip
 description ***UC1-CALLING-UC2***
 preference 1
 destination-pattern 640
 session target ipv4:10.1.2.1
 dtmf-relay h245-alphanumeric
!
dial-peer voice 2012 voip
 description ***UC1-CALLING-UC2***
 preference 1
 destination-pattern 641
 translate-outgoing called 600
 session target ipv4:10.1.2.1
 dtmf-relay h245-alphanumeric
!
dial-peer voice 2013 voip
 description ***UC1-CALLING-UC2***
 preference 1
 destination-pattern 642
 session target ipv4:10.1.2.1
 dtmf-relay h245-alphanumeric
!
dial-peer voice 2014 voip
 description ***UC1-CALLING-UC2***
 preference 1
 destination-pattern 643
 session target ipv4:10.1.2.1
 dtmf-relay h245-alphanumeric
!
dial-peer voice 2015 voip
 description ***UC1-CALLING-UC2***
 preference 1
 destination-pattern 698
 session target ipv4:10.1.2.1
 dtmf-relay h245-alphanumeric
 codec g711ulaw
!
dial-peer voice 622 voip
 destination-pattern 622
 translate-outgoing called 622
 session target ipv4:10.1.2.1
 codec g711ulaw
You may or May not want this DN but i tend to use it a lot and CCA puts it in as well
ephone-dn  86
 number 7... no-reg primary
 description ***CCA XFER TO VM EXTENSION***
 call-forward all 699

The above configuration does not rely on translation rules, however this was only chosen because the remote site does not have many end points and would not require a very large dial-peer configuration, which by the way you are limited to how many you can do as DP's are resource intensive.

How the above Works:

Well when someone in site A dials say 642, dial-peer 2013 will be matched (possible others but this would be the identical one or the closest one) and it will follow the instructions within the dial-peer configuration, a call will be sent to site B.

Site B:

voice call send-alert
voice rtp send-recv
!
voice service voip 
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 supplementary-service h450.12
 sip
  no update-callerid
!
!
!
voice class codec 1
 codec preference 1 g711alaw
 codec preference 2 g711ulaw
 codec preference 3 g729r8
voice translation-profile XFER_TO_VM_PROFILE
 translate redirect-called 2002
!
!
voice-card 0
 dspfarm
 dsp services dspfarm
This is site B's Voice Mail Pilot
dial-peer voice 50 voip
 destination-pattern 698
 b2bua
 session protocol sipv2
 session target ipv4:10.1.11.1
 dtmf-relay sip-notify
 codec g711ulaw
The following is a big dial-peer configuration as this remote site is talking back to the Head Office, but the resources could be afforded on this site
dial-peer voice 2011 voip
 description ***UC1-CALLING-UC2***
 preference 1
 destination-pattern 610
 session target ipv4:10.1.1.1
 dtmf-relay h245-alphanumeric
!
dial-peer voice 2012 voip
 description ***UC1-CALLING-UC2***
 preference 1
 destination-pattern 611
 session target ipv4:10.1.1.1
 dtmf-relay h245-alphanumeric
!
dial-peer voice 2013 voip
 description ***UC1-CALLING-UC2***
 preference 1
 destination-pattern 612
 session target ipv4:10.1.1.1
 dtmf-relay h245-alphanumeric
!
dial-peer voice 2014 voip
 description ***UC1-CALLING-UC2***
 preference 1
 destination-pattern 613
 session target ipv4:10.1.1.1
 dtmf-relay h245-alphanumeric
!
dial-peer voice 2015 voip
 description ***UC1-CALLING-UC2***
 preference 1
 destination-pattern 614
 session target ipv4:10.1.1.1
 dtmf-relay h245-alphanumeric
!
dial-peer voice 2016 voip
 description ***UC1-CALLING-UC2***
 preference 1
 destination-pattern 615
 session target ipv4:10.1.1.1
 dtmf-relay h245-alphanumeric
!
dial-peer voice 2017 voip
 description ***UC1-CALLING-UC2***
 preference 1
 destination-pattern 616
 session target ipv4:10.1.1.1
 dtmf-relay h245-alphanumeric
!
dial-peer voice 2018 voip
 description ***UC1-CALLING-UC2***
 preference 1
 destination-pattern 617
 session target ipv4:10.1.1.1
 dtmf-relay h245-alphanumeric
!
dial-peer voice 2019 voip
 description ***UC1-CALLING-UC2***
 preference 1
 destination-pattern 618
 session target ipv4:10.1.1.1
 dtmf-relay h245-alphanumeric
!
dial-peer voice 2020 voip
 description ***UC1-CALLING-UC2***
 preference 1
 destination-pattern 619
 session target ipv4:10.1.1.1
 dtmf-relay h245-alphanumeric
!
dial-peer voice 2021 voip
 description ***UC1-CALLING-UC2***
 preference 1
 destination-pattern 620
 session target ipv4:10.1.1.1
 dtmf-relay h245-alphanumeric
!
dial-peer voice 2022 voip
 description ***UC1-CALLING-UC2***
 preference 1
 destination-pattern 621
 session target ipv4:10.1.1.1
 dtmf-relay h245-alphanumeric
!
dial-peer voice 2023 voip
 description ***UC1-CALLING-UC2***
 preference 1
 destination-pattern 622
 session target ipv4:10.1.1.1
 dtmf-relay h245-alphanumeric
!
dial-peer voice 2024 voip
 description ***UC1-CALLING-UC2***
 preference 1
 destination-pattern 623
 session target ipv4:10.1.1.1
 dtmf-relay h245-alphanumeric
!
dial-peer voice 2025 voip
 description ***UC1-CALLING-UC2***
 preference 1
 destination-pattern 624
 session target ipv4:10.1.1.1
 dtmf-relay h245-alphanumeric
!
dial-peer voice 2026 voip
 description ***UC1-CALLING-UC2***
 preference 1
 destination-pattern 625
 session target ipv4:10.1.1.1
 dtmf-relay h245-alphanumeric
!
dial-peer voice 2027 voip
 description ***UC1-CALLING-UC2***
 preference 1
 destination-pattern 626
 session target ipv4:10.1.1.1
 dtmf-relay h245-alphanumeric
!
dial-peer voice 2028 voip
 description ***UC1-CALLING-UC2***
 preference 1
 destination-pattern 627
 session target ipv4:10.1.1.1
 dtmf-relay h245-alphanumeric
!
dial-peer voice 2029 voip
 description ***UC1-CALLING-UC2***
 preference 1
 destination-pattern 628
 session target ipv4:10.1.1.1
 dtmf-relay h245-alphanumeric
!
dial-peer voice 2030 voip
 description ***UC1-CALLING-UC2***
 preference 1
 destination-pattern 629
 session target ipv4:10.1.1.1
 dtmf-relay h245-alphanumeric
!
dial-peer voice 2031 voip
 description ***UC1-CALLING-UC2***
 preference 1
 destination-pattern 630
 session target ipv4:10.1.1.1
 dtmf-relay h245-alphanumeric
!
dial-peer voice 2010 voip
 description Incoming Call
 incoming called-number .
Some Examples of site B's DN's which might be helpful as you can see the call routing path
ephone-dn  10  dual-line
 number 640 secondary XXXXX640 no-reg both
 label Dept1-640
 description Dept1-640
 name Dept1-640
call-forward busy 698
call-forward noan 698 timeout 10
!
!
ephone-dn  11  dual-line
 number 641 secondary XXXXX641 no-reg both
 label Dept2-641
 description Dept2-641
 name Dept2-641
 call-forward busy 698
 call-forward noan 698 timeout 10
!
!
ephone-dn  12  dual-line
 number 642 secondary XXXXX642 no-reg both
 label Dept3-642
 description Dept3-642
 name Dept3-642
 call-forward busy 698
 call-forward noan 698 timeout 10

Some Notes: Site A and Site B work within a private network so do not use the above IP address ranges unless it falls within your networks subnet's, make sure these are changed accordingly. You will also note the lack of translation rules, whilst you can run them as well as the dial-peers it is recommended you choose one of them, if there is a heap of extensions and this amounts to a lot of dial-peers then you might need to use translation rules and dial-peer 2010 configuration.

Site B also has Direct Inward Dial numbers that are controlled from Site A, you might not have that but in this scenario you can use it it is quite useful if you want to give them DID's.

Ok the above is most likely not the recommended deployment of it, this was a make do configuration but it worked so it stayed that way

Answer to your question:

One question for the understanding :
If extension 15 on node 301 calls extension 33 on node 501
and this extension is forwarded to 55 (the CUE on node 501)
does node 501 inform node 301 about this?
And starts node 301 then a new call to extension 55 on node 501?
In this case, node 301 should dial 50155 to reach the CUE on the
other side, but how does the CUE on node 501 know to which
mailbox this call from 30115 is adressed?

Short answer is NO, Site-A only cares about itself and the rules applied to your configuration, if you have a CF-all to VM rule on a DN, then it will just follow that rule, Site-B wont know about this until it hits the VM, although there will be some background chatter that will take place, but the bulk of the chatter will be on the receiving end.

Brief Comment:

Remarks:
Node 501 is configured only with one subnet for UC, CUE and Phones
because the customer want to reach everything from remote without
NAT/PAT and he has no chance to change the routes in the MPLS by himself.
So I´ve got only a /27 to set the whole box up; it took me several hours, but it works.
Node 301 is an old system, just for the lab and it is used for every testings,
so there are many confusing config-parts.

Flat Network deployments are common when they reside on MPLS networks, I have only ever worked with 3 of them myself but as long as you have everything configured right with the routing, then all should be fine. NOTE: I am not a data expert, in fact I struggle with it a bit and rely on those around me who are experienced in it, I just love working with voice.

By the way, if you are running 2 digit extensions, can I convince you to go to 3 digit extension dialing? I wont go into why it is better now as that is an entirely different subject, but 3 or 4 digit dialing is the best to work with.

Ok eyes officialy hanging out of my head now, I need a cold beer and then urgently needed sleep.

Honestly the above is just a guide it may or may not help you, just give it your best shot if it doesnt work then maybe when I am in a better state i can try to help you out more.

Good luck and I wish you all the best, my apologies about the long post but heres to hoping it helps you out

Cheers,

David.