Dial-Peer configuration help in CME

[sampath9614]

Hello experts,

i have been trying to get my voice network up from 2days but no luck.

Here is my scenario:

CUCM

CME

Iris PBX

+++++++++++++++Config posted at the Bottom+++++++++++++++++

CME and PBX are PRI trunked; i can see layer 1 active and layer 2 multi_Frame_Established in "show isdn status".

SIP trunk from CUCM to CME Established.

Unable to communicate from Voip phones registered to CUCM to analog phones in PBX.

Few times voip to analog calls working and then few changes in the PBX making it crap again. (Those were made by PBX technician)

Voip ext: 1003

Analog Ext: 270

!!for voip to analog calls

In CUCM:

Route pattern: 9XXX --> to CME trunk

In CME:

Destination pattern: 9... --> to E1 card (with auto stripping)

!!for analog to voip calls

In PBX the guy given access code as 4 and stripped it while sending.

that means from analog ext to reach 1003 user will dial 41003 and 4 will be stripped off.

In CME:

Destination pattern: 1... -->  session target ipv4:<ip of CUCM>

With this configuration in the "debug ccapi inout" am seeing "Transfer number is NULL" and

in the "show call history last 1" am seeing cause code "1C" -- Invalid number.

After doing some research these are my ideas but no tested:

1. I should have incoming dial peer to match traffic from e1 card:

dial-peer voice X pots

incoming called-number .T

2) I should configure dtmf-relay sip-notify and dtmf-relay rtp-nte for accepting digits from telephone-service.

3) Do i need create any DIDs for incoming calls from pbx?

Please help me on this, how could i get calls from analog to voip and vice versa, what i am thinking is right? if not what could be the alternatives? where we should use dtmf-relay commands.?

Thanks everyone in advance.

Existing config:

card type e1 0 1
network-clock-participate wic 1
network-clock-select 1 E1 0/1/1
isdn switch-type primary-qsig
voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 no supplementary-service h450.2
 no supplementary-service h450.3
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 sip
  registrar server expires max 600 min 60
  no update-callerid
!
voice class codec 1
 codec preference 1 g729br8
 codec preference 2 g729r8
 codec preference 3 g723ar63
 codec preference 4 g711ulaw
 codec preference 5 g711alaw
!
controller E1 0/1/1
 framing NO-CRC4
 pri-group timeslots 1-31
!
voice-port 0/1/1:15
 bearer-cap Speech
mgcp
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
!
!
!
dial-peer voice 3 pots
 destination-pattern 9...
 port 0/1/1:15
!
dial-peer voice 1 voip
 destination-pattern 1...
 session target ipv4:10.15.108.242
!
!
!
!
gatekeeper
 shutdown
!
!
telephony-service
 max-conferences 8 gain -6
 transfer-system full-consult
!
!

[Robert Shaw]

The CME box is not seeing 9501. The leading 9 is being stripped by the ISDN before it gets to the CME box.  If thats the case you will need to modify the Dial-Peer voice 3 voip to 5..

At this point it fails to select a Dial-peer so will not forward anything the CUCM.

[Robert Shaw]

I'm getting a little confused.  I was under the impression 270 was an analog phone coming inbound over the ISDN to an IP Phone registered on CUCM beginning with 1...  If you are calling 9501 from 270 the call is going to hairpin out of the ISDN line, meaning it will come in and go out of the ISDN, is this what you want to do?

Could you also post the output of the following?

show dial-peer voice summary

Thanks
Rob

[sampath9614]

i have changed the dial peer accordingly.. please see the attached.

source 270 analog ext

dest 501 voip phone registered to cucm

no css and partitions.. voip to analog still working.. but not return calling.

[Robert Shaw]

Now you have made the change can you retest with the same debugs and post the output?

[sampath9614]

those logs were taken after changing the dial peers only... when i have changed the dn in cucm then changed the DPs after changing took the debug

[Robert Shaw]

I don't see the debugs I only see the picture of your screen showing the Dial-Peers.

[sampath9614]

please see the attached..

[Robert Shaw · Accepted solution]

Ok, Lets remove all of your Dial-Peers and start from scratch.

For testing purposes we will test with specific numbers.  We can get more general later.  Try the following for inbound.

dial-peer voice 10 pots

desc ## Inbound from Iris PBX ##

incoming called-number .

 direct-inward-dial
 forward-digits all

Then lets try this for your Call Manager Dial-Peer

dial-peer voice 20 voip

destination-pattern 501

session protocol sipv2

session target ipv4: CUCM_IP

Please retest and submit fresh outputs showing,

debug ccapi voice inout

debug ccsip messages

debug isdn q931

[sampath9614]

from analog phone when i press the access code 9, i.e., 

analog ext: 270

voip ext: 1003

pbx guyz gave 9 as the trunk for pri, when pressing the digit 9 it's giving busy tone and showing 

Cause i = 0x809C - Invalid number format (incomplete number)

[sampath9614]

It just worked bro... 

Think bpx side.. 91.. was assigned for another pattern.

By changing 501 it worked properly.

Now everything is working as expected. 

Whole credit to you only bro.. thank you very much.

** Saved my life.