Call-forward all to SIP CME 15 Issue

[esacco]

Hello, 

I have a problem when, from a SCCP phone I enable the call-forward all to another SIP extension (only SIP extension), the error in debug log is "Cause i = 0x80AF - Resource unavailable, unspecified.". This happen both from internal and external calls. If I try to set call-forward in "voice register dn" it work as the SCCP to SIP (and vice versa) call transfer.

This is my config:

!

voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
address-hiding
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
no supplementary-service sip handle-replaces
supplementary-service media-renegotiate
supplementary-service ringback h225-info
redirect ip2ip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
h323
modem passthrough nse codec g711alaw
sip
header-passing
registrar server expires max 1200 min 60
no update-callerid
!

voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8

!

dial-peer voice 1000 voip
description *** Trunk IP DECT interni 65x ***
destination-pattern 65.
b2bua
session protocol sipv2
session target ipv4:192.168.100.201:5060
session transport udp
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte sip-notify
no vad
supplementary-service pass-through
!

!

telephony-service
video
ip source-address 192.168.100.200 port 2000
caller-id block code *9
calling-number initiator
service phone videoCapability 1
service phone ehookenable 1
service phone webAccess 0
service dnis overlay
service dnis dir-lookup
timeouts interdigit 5
system message Cuvage
url directories http://192.168.65.100/fw/apps/Speedy/xml/directories/default.aspx
url services http://192.168.65.100/fw/Apps/Ivory/xml/overrides IVR
url authentication http://192.168.65.100/fw/authenticate.asp
cnf-file location flash:
cnf-file perphone
user-locale IT load CME-locale-it_IT-Italian-11.7.11.7.tar
network-locale IT
load 7916-12 B016-1-0-4
load 7916-24 B016-1-0-4
load 7911 SCCP11.9-4-2SR3-1S
load 7941 SCCP41.9-4-2SR3-1S
load 7945 SCCP45.9-2-1S
load 7965 SCCP45.9-2-1S
load 6921 SCCP69xx.9-4-1-3SR3
load 6941 SCCP69xx.9-2-1-0
load 6961 SCCP69xx.9-2-1-0
time-zone 23
time-format 24
date-format dd-mm-yy
voicemail 999
max-conferences 8 gain -6
call-park system application
call-forward pattern .T
moh Buble.au
multicast moh 239.1.1.1 port 16384
transfer-system full-consult
transfer-pattern .T
log password 
secondary-dialtone 0
xmltest

!

Thank you

[Maren Mahoney]

Can you clarify the problem a little? What I'm getting is that if you CfwdAll from the SCCP extension (ephone-dn) to a SIP extension (voice register dn), inbound calls to the ephone-dn fail to transfer with a "resource unavailable" error. But if you CfwdAll from the SIP extension to the SCCP extension it works fine. Do I have that right?

My first thought is that the router is looking for a transcoder. By default, because SIP CME directory numbers create virtual voip dial peers (in the 40000 range) the default codec used by a SIP CME phone is g729. And since SCCP CME directory numbers create virtual pots dial peers (in the 50000 range) they use g711/g722 by default.

Is the 'voice class codec 1' also assigned to both the ephone-dn and the voice register dn? If not, try assigning that to the SIP CME number (and the ephone-dn if you want to be safe) and see if that fixes the problem.

[m.minetti]

Hi, you understood, when CfwdAll to a SIP extension is enabled the SIP phone fail to transfer with a "resource unavailable" error.

The costumer have an old 2851 Call manager express with the last firmware available (151-4.M12a) and in this release I can't find "voice-class codec" command under ephone-dn and voice register dn, I can configure "voice-class codec" only in voice register pool. Yesterday I tried to configure SNR service in a one extension and work fine with this workaround.

Thank you

[Maren Mahoney]

My bad. Yes, for the SIP CME it is under voice register pool. Did you try assigning that to the voice register pool to see if that solved the problem? I'm glad you found a work-around, but I'd like to help you solve the underlying problem.

[fedor.solovev]

Hi, how did you solve the problem ?