CME 4.1, Call Forward All doesn't work (SIP Provider)

alig.norbert · ‎03-02-2007

Hi all,

I had a previous problem with an CCME 4.0(0).

http://forum.cisco.com/eforum/servlet/NetProf?page=netprof&forum=Unified%20Communications%20and%20Video&topic=IP%20Telephony&CommCmd=MB%3Fcmd%3Dpass_through%26location%3Doutline%40%5E1%40.1ddd38eb

After upgrading to CME 4.1(0) (c2801-ipvoicek9-mz.124-11.XJ.bin) the "call transfer" (on the IP-Phone) from an incoming SIP-Call to an external number through the same SIP-Provider is working.

The "call forward all" instead build up a SIP-signaling channel (ringing on the external phone) but no RTP-flow.

(ephone-dn 6)

Here the partial config:

...

....

voice-card 0

!

voice call send-alert

voice call convert-discpi-to-prog

voice rtp send-recv

voice disc-pi-incoming-on

!

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 sip moved-temporarily

no supplementary-service sip refer

redirect ip2ip

h323

h245 caps mode restricted

sip

!

voice class codec 1

codec preference 1 g711alaw

codec preference 2 g711ulaw

!

voice class h323 1

call start slow

no call preserve

!

voice translation-rule 40

rule 1 /\(.*\)/ /0\1/

!

voice translation-rule 190

rule 1 /^0\(.*\)/ /\1/

rule 2 /^9\(.*\)/ /\1/

!

voice translation-rule 191

rule 1 /^2\([0-4].\)/ /07192099\1/

rule 2 /^7\([0-4].\)/ /07192099\1/

!

voice translation-rule 192

rule 2 /^07192099\(..\)/ /2\1/

rule 10 /.*/ /200/

!

voice translation-rule 220

rule 2 /^0\(.*\)/ /9\1/

!

voice translation-profile TP_IN_SIP

translate calling 40

translate called 192

!

voice translation-profile TP_OUT_SIP

translate calling 191

translate called 190

!

dial-peer voice 200 voip

description *** SIP-TRUNK (OUT/IN) ***

translation-profile incoming TP_IN_SIP

translation-profile outgoing TP_OUT_SIP

max-conn 4

destination-pattern 0.T

redirect ip2ip

max-redirects 4

session protocol sipv2

session target ipv4:21x.xx.198.140

session transport udp

incoming called-number 07192099..

dtmf-relay rtp-nte

codec g711alaw

no vad

gateway

timer receive-rtp 1200

!

sip-ua

authentication username xxx password xxxxxxx

retry invite 1

retry response 1

retry bye 1

retry register 1

retry options 0

registrar ipv4:2xx.xx.198.140 expires 60

telephony-service

load 7960-7940 P0030702T023

load 7914 S00104000100

load ATA ATA030203SCCP051201A.zup

max-ephones 30

max-dn 150

ip source-address 192.168.115.1 port 2000

calling-number local

timeouts interdigit 4

system message TEST

url services http://192.168.115.1/localdirectory

user-locale DE

network-locale CH

time-zone 23

time-format 24

date-format dd-mm-yy

keepalive 45

max-conferences 4 gain -6

call-forward pattern .T

moh music-on-hold.au

web admin system name xxx password xxx

dn-webedit

time-webedit

transfer-system full-consult

transfer-pattern .T

secondary-dialtone 0

ephone-dn 6

number 206 no-reg primary

label Temporaer(206)

description 071/920.99.06

name Temporaer

call-forward all 0079412345

ephone-dn 150 => for sip-ua registration

number 0719209900

See attached debug voip rtp output.

Any ideas ?

Thanks & greets,

Norbert

danm · ‎12-30-2007

I'm having the same problem... Any resolution? Thanks.. It would be a big help.

paolo bevilacqua · ‎12-30-2007

Hi,

if you are diverting an IP call to another IP leg, that requires IP-to-IP GW images, the ones that end with _ivs.

danm · ‎12-30-2007

I'm running c3745-adventerprisek9-mz.124-11.XW5.bin

I just have a 3745 with a SIP trunk... If I select an extension and press Call-Forward all to another PSTN number it doesn't work (Get a recording that states " The number you are calling is not in service"

paolo bevilacqua · ‎12-30-2007

Ok, I may have misunderstood the call flow.

Can you restate it again ?

If you get such a recording, one way or another, the wrong is being called.

danm · ‎12-30-2007

Thanks in advance for you help, I really appreciate it...

I have 3745 CME 4.2 running c3745-adventerprisek9-mz.124-11.XW5.bin with a SIP Provider.

If somecalls a DID number, it rings the phone, I can then transfer it internally, or to the PSTN. Works great.

If somecalls a DID number that has CFA to an internal number it works. Here is the problem, If its set to CFA to a PSTN number. The calling party hears "Thenumber you are calling is not in service".

I did a debug voice ccapi, and verified that both numbers are showing correctly... I have no supplemental-service sip moved-temp and no supplemental-service sip refer.

I'm stumped... Is it not hairpinning or can i force it to keep both legs up, instead of dropping out of the call flow.

paolo bevilacqua · ‎12-30-2007

Ok. Please collect "debug ccsip message" for the purpose of verifying the numbers again. I suspect an _ivs image will be required after all, but you should not get such message in first place.

alig.norbert · ‎12-31-2007

Hi

I suppose the message "The number you are calling is not in service" is from the PSTN. Try the CFA without the PSTN-prefix.

Here my partial config:

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 sip moved-temporarily

h323

h245 caps mode restricted

sip

voice translation-profile TP_IN_SIP

translate calling 40

translate called 192

!

voice translation-profile TP_OUT_SIP

translate calling 191

translate called 190

voice translation-rule 40

rule 2 /\(.*\)/ /0\1/

!

voice translation-rule 190

rule 1 /^0\(.*\)/ /\1/

!

voice translation-rule 191

rule 1 /^2\([0-4].\)/ /07192099\1/

rule 2 /^7\([0-4].\)/ /07192099\1/

!

voice translation-rule 192

rule 2 /^0719209\(..\)/ /2\1/

dial-peer voice 210 voip

description *** SIP-TRUNK (IN) ***

translation-profile incoming TP_IN_SIP

translation-profile outgoing TP_OUT_SIP

max-conn 2

destination-pattern 07192099..

session protocol sipv2

session target ipv4:212.xxx.xxx.140

session transport udp

incoming called-number 07192099..

dtmf-relay rtp-nte

codec g711ulaw

no vad

!

dial-peer voice 200 voip

description *** SIP-TRUNK (OUT) ***

translation-profile incoming TP_IN_SIP

translation-profile outgoing TP_OUT_SIP

max-conn 2

destination-pattern 0.T

session protocol sipv2

session target ipv4:212.xxx.xxx.140

session transport udp

incoming called-number 07192099..

dtmf-relay rtp-nte

codec g711ulaw

no vad

!