03-02-2007 12:49 AM - edited 03-14-2019 08:17 PM
Hi all,
I had a previous problem with an CCME 4.0(0).
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
12-30-2007 01:57 PM
I'm having the same problem... Any resolution? Thanks.. It would be a big help.
12-30-2007 02:46 PM
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.
12-30-2007 02:54 PM
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"
12-30-2007 02:59 PM
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.
12-30-2007 03:06 PM
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.
12-30-2007 03:17 PM
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.
12-31-2007 09:21 AM
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
!
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