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

Unanswered Question
Mar 11th, 2011

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

I have this problem too.
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adcompto Mon, 03/14/2011 - 08:49

What kind of phone lines do you have?  If you have a SIP trunk,  The provider possibly doesn't like the SIP diversion header.

Regards

Adam Compton

DetNit1101 Tue, 03/15/2011 - 00:03

Hi,

at each UC there are two ISDN-lines from the PSTN

and the trunk/link between the UCs is within an MPLS-cloud.

(in the moment the MPLS-coud is represented by a CISCO 1712)

No SIP ...

Kind regards, DetNit

adcompto Tue, 03/15/2011 - 05:42

hmmm. interesting....

I wonder if registering a transcoder or MTP resource would make any difference.  Of course. this is well beyond CCA configuration, but it looks like you did the dial-peers manually.

Adam Compton

DetNit1101 Tue, 03/15/2011 - 05:47

Hi Adam,

this is right - no CCA used for this configuration.

The CLI is my buddy and I took some configuration guidelines from the CME.

I´m confused about this dysfunction and have no further ideas.

Kind regards,

DetNit

adcompto Tue, 03/15/2011 - 05:55

A transcoder is always worth a shot.  Here is the configuration guide for transcoder on CME:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmetrnsc.html

basically looks like this

sccp local vlan1

sccp ccm 10.1.1.1 id 1 priority 1 version (5.0 or above).

sccp

sccp ccm group 1

associate ccm 1 pri 1

associate profile 1 reg uc500xcode

dspfarm profile 1 transcode

codec g711u

codec g711a

codec g729r

codec g729br

max sessions 5

associate application sccp

no shut

telephony-service

sdspfarm units 1

sdspfarm transcode 5

sdspfarm tag 1 uc500xcode

This is just an example and would need to be altered to match your system.  I figured i would would type something up for you real quick

Regards,

Adam Compton

David Trad Tue, 03/15/2011 - 16:15

Hi DetNit,

Before you apply that Config can you check please to see if you have enough DSP resources to do it? I have seen machines go bad when applying the config and there was not enough resources for it.

ROUTER# sh dspfarm all

And it should look something similar to this:

Dspfarm Profile Configuration

Profile ID = 3, Service = TRANSCODING, Resource ID = 1 

Profile Description : Transcoding when conferencing is enabled

Profile Service Mode : Non Secure

Profile Admin State : UP

Profile Operation State : ACTIVE IN PROGRESS

Application : SCCP   Status : ASSOCIATION IN PROGRESS

Resource Provider : FLEX_DSPRM   Status : UP

Number of Resource Configured : 1

Number of Resource Available : 1

Codec Configuration

Codec : g711ulaw, Maximum Packetization Period : 30

Codec : g711alaw, Maximum Packetization Period : 30

Codec : g729ar8, Maximum Packetization Period : 60

Codec : g729abr8, Maximum Packetization Period : 60

Codec : g729r8, Maximum Packetization Period : 60

Codec : g729br8, Maximum Packetization Period : 60

Dspfarm Profile Configuration

Profile ID = 1, Service = CONFERENCING, Resource ID = 2 

Profile Description : Conferencing profile

Profile Service Mode : Non Secure

Profile Admin State : UP

Profile Operation State : ACTIVE

Application : SCCP   Status : ASSOCIATED

Resource Provider : FLEX_DSPRM   Status : UP

Number of Resource Configured : 2

Number of Resource Available : 2

Codec Configuration

Codec : g711ulaw, Maximum Packetization Period : 30 , Transcoder: Not Required

Codec : g711alaw, Maximum Packetization Period : 30 , Transcoder: Not Required

SLOT DSP VERSION  STATUS CHNL USE   TYPE    RSC_ID BRIDGE_ID PKTS_TXED PKTS_RXED

0    2   23.8.0   UP     N/A  FREE  xcode  1      -         -         -       

0    4   23.8.0   UP     N/A  FREE  conf   2      -         -         -       

0    4   23.8.0   UP     N/A  FREE  conf   2      -         -         -       

Total number of DSPFARM DSP channel(s) 3

It is important to know what you have available before you make any configuration changes... OH and please please please make a backup of your config before you make the changes, and put that backup on the flash memory, even if you do not commit the changes.

Cheers,

David.

DetNit1101 Wed, 03/16/2011 - 02:17

Hi David,

thank you for this information.

Below the output, it´s identical on the second UC.


MOSKAU_VOIP1#sh dspfarm all
DSPFARM Configuration Information:
Admin State: DOWN, Oper Status: DOWN - Cause code: ADMIN_STATE_DOWN
Transcoding Sessions: 0(Avail: 0), Conferencing Sessions: 0 (Avail: 0)
Trans sessions for mixed-mode conf: 0 (Avail: 0), RTP Timeout: 600
Connection check interval 600 Codec G729 VAD: ENABLED

Total number of active session(s) 0, and connection(s) 0


Total number of DSPFARM DSP channel(s) 0

MOSKAU_VOIP1#

and if I try this I get following answer

MOSKAU_VOIP1(config)#dspfarm profile 1 transcode universal


Dspfarm profile 1 :: No resource, check voice card or dspfarm service is not configured
MOSKAU_VOIP1(config-dspfarm-profile)#

How can I setup this resources?

Kind regards,

DetNit

adcompto Wed, 03/16/2011 - 05:24

I forgot to add this because it is normally setup on a uc500:

voice-card 0

dsp service dspfarm

You should be able to program a dspfarm profile.

Adam Compton

DetNit1101 Wed, 03/16/2011 - 07:16

Hi Adam,

I´ve done this configuration


voice-card 0

dsp service dspfarm

but I´m not able to give a maximum session > 0

and so I can´t make a no shut, because it´s need a value >0

Kind regards,

DetNit

adcompto Wed, 03/16/2011 - 07:31

You might have maxed out your dsps.  What kind of UC500 do you have?

David Trad Wed, 03/16/2011 - 15:49

Hi DetNit,

I would really love to log into your system (Via SSH) and have a look at this, even based on it being a 2 BRI or even a 4 BRI system you should still have enough DSP resources to at least do minimal conferencing.

Something is not right and I would really love to investigate this further.

Cheers,

David.

DetNit1101 Thu, 03/17/2011 - 04:11

Hi,

i took the config and put it on a UC540W-BRI-K9 and the behaviour is the same.

I put the config from Adam on the box and I was able to set a maximum-connection value, but nothing changed.

No informaton with several "deb dsp..."-commands

I think that is not a problem of not enough DSPs ...

The only information with deb voice ccapi error on incomig calls

(but what´s the meaning of this?)


000143: Mar 17 12:08:06.175: //-1/xxxxxxxxxxxx/CCAPI/cc_validate_mlpp_info:
   Unsupported MLPP Service Domain Network 0
000144: Mar 17 12:08:06.175: //-1/xxxxxxxxxxxx/CCAPI/cc_validate_mlpp_info:
   Unsupported MLPP Service Domain Network 0

SHIT! (sorry for this ...)

Kind regards,

DetNit

adcompto Thu, 03/17/2011 - 05:07

MLPP would not cause the call to fail.

I don't really have any other ideas without looking at show run for both sites, and debug voip ccapi inout for both sites on one call.

Adam Compton

DetNit1101 Thu, 03/17/2011 - 05:59

Hi Adam,

i´ve done the debugs:

The call was generated from Node 301 extension 15 to node 501 extension 33,

the extension 15 on node 501 was forwarded to voicemail 55.

I´ve dialed 50133 on node 301.


The node 501 has a special configuration:

All IPs in one subnet, so I configured the box in CME-Style and normal function is given

and the EXPANSION-Port Fa0/1/8 is my WAN-port, WAN-Port Fa0/0 is unused.

(Guideline from the cusctomer)

The problem exits although from a call from node 501 extension 33 to node 301 extension 15.

The second UC is my lab-UC with some other pieces of configurations from other testings/implementations.

The connection between the subnets is realized by a cisco1712.

Thanks in advance,

DetNit

adcompto Thu, 03/17/2011 - 05:17

I figured out what the transcoder issue is.  You have to configure all of your codecs before you configure max sessions.  I tried it here in the lab on a 540 and I got the same error you got.

Adam Compton

PS  Also can you show me what your translation-profiles/rules look like?

DetNit1101 Thu, 03/17/2011 - 06:40

Hi,

I´ve tried the steps again and got this message, but it is still not working ...


Node_501(config-sccp-ccm)#
000672: Mar 17 13:22:54.283: %SDSPFARM-6-REGISTER: mtp-1:uc500xcode IP:10.212.10.198 Socket:7 DeviceType:MTP has registered.
Node_501(config-sccp-ccm)#

sdoherty Thu, 03/17/2011 - 07:50

Had the same issue here and had to ADD a diversion header to all the outbound calls..

request INVITE sip-header Diversion add "Diversion:;privacy=off;reason=unconditional;screen=no"

adcompto Thu, 03/17/2011 - 07:14

outgoing from node 301 seems fine.

on node 501, it matches and incoming dial-peer;

----- ccCallInfo IE subfields -----
   cisco-ani=30115
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=0
   dest=50133
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=-1
   cisco-rdnplan=-1
   cisco-rdnpi=-1
   cisco-rdnsi=-1
   cisco-redirectreason=-1   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0

000462: Mar 17 13:34:40.567: //-1/60FF55C480E3/CCAPI/cc_api_call_setup_ind_common:
   Interface=0x87418D7C, Call Info(
   Calling Number=30115,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=50133(TON=Unknown, NPI=Unknown),
   Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
   Incoming Dial-peer=90, Progress Indication=ORIGINATING SIDE IS NON ISDN(3), Calling IE Present=TRUE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=11

The problem is, it doesn't match an outgoing dial-peer.  You are using  translation-profile incoming ankommend_von_FRLHV


the rule that it should match is :

voice translation-rule 91
rule 1 /^...\(..\)/ /\1/

I don't see anything particularly wrong with it, but I don't think it is being translated, and it is definitely not matching the outgoing dial-peer voice extension 33.  Normally I would see in the debugs something similar to the above with the translated number and an outgoing dial-peer listed.  Even if it is going to an ephone-dn, there would still be an outgoing dial-peer.

You might want to try using a more specific translation-rule:

rule 1 /^501\(..\)$/ /\1/

Adam

DetNit1101 Thu, 03/17/2011 - 07:44

Sorry,

I thaught I can use dial-peer 90 for both direction in or out,

so I gave a destination-pattern for outgoing an a incoming

dialed number to recognize incoming calls.

And for every direction a seperate translation rule, incoming or outgoing?

Why do I need to use an outgoing rule when a call comes in?

Have I got a wrong understanding about how it works?

I´m from the router world, so I think in/out is similar to access-lists :-)

DetNit

adcompto Thu, 03/17/2011 - 08:01

You can use the dial-peer in or out, but the destination pattern is 301.. which will not match.  This is one reason I suggested separating the incoming and outgoing dial-peers because it gives you more granularity into what is incoming and what is outgoing.  Ephone-dns also have an associated dial-peer that is created in the background.

This call should match 90 incoming, but should match the dial-peer for ephone-dn with number 33 for the outgoing.

DetNit1101 Thu, 03/17/2011 - 08:21

ok, so far so good

But if the extension is not forwardes, so I can reach it.

An outgoing dial-peer for extension 33 should exits.

What´s about the statement in the debug from Node501


000478: Mar 17 13:34:40.571: //-1/xxxxxxxxxxxx/CCAPI/ccUpdateRedirectNumber:
   type=6  Original Called Number=33, Called Number=33, Calling Number=30115, Calling DN=-1 Calling Id=11,
   Redirect Number=55, Redirect Reason=15
000479: Mar 17 13:34:40.571: //-1/xxxxxxxxxxxx/CCAPI/ccUpdateRedirectingNumber:
   type=6 redirecting_number=33

Is there something going wrong?


DetNit

adcompto Thu, 03/17/2011 - 08:43

Can you post the debug of a working call that is not forwarded?

adcompto Thu, 03/17/2011 - 09:59

Can you try removing supplementary services from the dial-peers?

DetNit1101 Thu, 03/17/2011 - 10:21

done, without any result ...

I´ve tested with deb voice dialpeer all on both sides.

It seems, that node 501 tells node 301 to call the redirected number 55,

but node 301 doesn´t make it well and found no dial-peer

I´ve added on node 301

!
voice translation-profile ankommend_von_DEWOB
translate called 72

translate redirect-called 73

translate redirect-target 73

!
voice translation-rule 73
rule 1 /^\(..\)/ /501\1/


and the corresponding part on node 501, but no effort

Today it´s enough, it´s already 6:00 pm.

Hope to get you tomorrow!!!

Until yet:  Many thanks!!!!

DetNit

David Trad Thu, 03/17/2011 - 15:38

Hi DetNit,

When I do a config comparison I find the following missing, they may not be relevant but something worth while looking into

This is based on the 501 Config....

You have:
stcapp ccm-group 1

Comparison Config Has:

stcapp ccm-group 1

stcapp

!

stcapp feature access-code

You do not have:

in "Voice Service Voip":

supplementary-service h450.12

You have:

voice class codec 1

codec preference 1 g711alaw

codec preference 2 g711ulaw

Comparison Config has:

voice class codec 1

codec preference 1 g711alaw

codec preference 2 g711ulaw

codec preference 3 g729r8

You do not have:

voice class cause-code 1

no-circuit

You do not have:

voice-card 0

dspfarm

dsp services dspfarm

You have:

sccp ccm 10.212.10.199 identifier 1 version 3.1

!

sccp ccm group 1

associate ccm 1 priority 1

Comparison Config has:

sccp local BVI101

sccp ccm 10.1.2.1 identifier 2 version 7.0

sccp

!

sccp ccm group 1

bind interface BVI101

associate ccm 2 priority 1

associate profile 1 register xcode01

!

dspfarm profile 1 transcode 

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

maximum sessions 2

associate application SCCP

You have:

telephony-service

video

fxo hook-flash

max-ephones 14

max-dn 56

ip source-address 10.212.10.198 port 2000

auto assign 1 to 1 type bri

calling-number initiator

service phone videoCapability 1

service phone ehookenable 1

service dnis overlay

service dnis dir-lookup

service dss

timeouts interdigit 5

system message XXXXXXXXXXXX

url services http://10.212.10.199/voiceview/common/login.do

time-zone 23

keepalive 30 auxiliary 4

voicemail 55

max-conferences 8 gain -6

call-forward pattern .T

call-forward system redirecting-expanded

moh flash:/media/music-on-hold.au

multicast moh 239.10.16.16 port 2000

dn-webedit

time-webedit

transfer-system full-consult

transfer-pattern .T

secondary-dialtone 0

fac standard

create cnf-files version-stamp 7960 Mar 11 2011 09:19:07

Comparison Config has:

telephony-service

sdspfarm units 1

sdspfarm transcode sessions 2

sdspfarm tag 1 xcode01

video

em logout 0:0 0:0 0:0

max-ephones 22

max-dn 88

ip source-address 10.1.2.1 port 2000

max-redirect 20

auto assign 10 to 27

auto assign 5 to 8 type anl

calling-number initiator

service phone videoCapability 1

service dnis overlay

service dnis dir-lookup

timeouts interdigit 5

system message XXXXXXXXXXXXXXXXX

url services http://10.1.11.1/voiceview/common/login.do

url authentication http://10.1.11.2/CCMCIP/authenticate.asp 

cnf-file location flash:

cnf-file perphone

time-zone 47

voicemail 698

max-conferences 8 gain -6

call-forward pattern .T

call-forward system redirecting-expanded

moh music-on-hold.au

multicast moh 239.10.16.16 port 2000

dn-webedit

time-webedit

transfer-system full-consult dss

transfer-pattern 9.T

transfer-pattern .T

transfer-pattern 0.T

transfer-pattern 6... blind

secondary-dialtone 0

night-service code *99

Now I have removed some sensitive data that should not be posted up on public, this will be represented as spaces in the config, also IP addresses will be totally different but this should be expected due to the differences in the systems, but the general idea should be there.

To me there are quite a few important things missing, things I would personaly never leave out on a CLI based config, and which CCA would make sure is on the system.

As I said before they may or may not be related, but to me they are as the config I compared it to is based on a SITE-2-SITE configuration, and both of them are a direct replication of each other minus the number ranges in use, to me this is the only way to do it, and it does not matter if they systems are different models, the config should be consistent across them both

I ran your config up on 2 UC-520 systems (Had to loan them as I don't have them myself) and found major issues with your 301 and 501 configuration, heaps of config changes had to be made to try and get it to work, so I am surprised yours is even working at all

My Advise is to blow the config away, use CCA 3 and redo the whole thing again, estimated time would be 3-4 hours per system but in my opinion a truck load quicker and less time resource intensive then diagnosing and debugging the current configuration.

Just my couple of cents worth, and I couldnt spend anymore time on it otherwise I would have just gone the next step further and re-written the whole thing for you

Cheers,

David.

DetNit1101 Fri, 03/18/2011 - 02:13

Hi David,

lot of thanks for your work!

The posted configs are edited several times by testing different issues, so they are really ugly.

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?

The debugs looks like this behaviour and I was really confused.

I thought, node 501 will handle this forwarding on his own.

I will try to clean both nodes and set them up straight but I try to do this without CCA,

because at the original customer side, the UC is connected to two CME.

Unfortunately I do not have any CME (I´m only the guy for the low-budget-segment).

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.

Kind regards,

DetNit

I followed your hint and have deleted the configs of the nodes from the forum.

DetNit1101 Fri, 03/18/2011 - 05:09

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 Fri, 03/18/2011 - 05:59

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.

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