01-04-2011 11:16 AM - edited 03-16-2019 02:41 AM
I'm getting this error message (Can Not Complete Conference) on the following scenario:
CME with local IP Phones with a SIP Trunk to a provider.
First, I start a local call (between my extensions), and then I pŕess "Confrn"
I start a second call (external one this time) and when I press the "Confrn" button againt to start a conference, I get this error message on my IP Phone : "Can Not Complete Conference" .
"debug ephone detail" gave me the following information:
Jan 4 19:16:30.162: called DN -1 chan 1, calling DN -1 chan 1 phone 11 s2s:0
Jan 4 19:16:30.162: ephone-11[10/5][SEP00127F7F9468]:DisplayMessageTag: tag 138 (Can Not Complete Conference)
Jan 4 19:16:30.162: ephone-11[10/5]:Failed to add DN 7 chan 1 to conference with conf_dn -1 chan 1
Ephone-11 is the one I'm using to start the conference.
Any ideas would be helpful.
Regards,
le_brito
01-04-2011 01:02 PM
Hi le_brito,
I suppose you must be using g711 for the internal calls. Is is true? Which codec are you using for the external caller over the SIP trunk? Are the internal conference calls working fine? Is the conference call failing only when the external caller is over a sip trunk or even for other PSTN users? Pl try all these and post the results.
Also try
ephone-dnocto-line
conference ad-hoc
Post a sh tech of your CME here for further troubleshooting.
Regards
Nitesh
PS: Pl rate helpful posts.
01-04-2011 02:07 PM
That's correct, I´m using g711 for internal calls and g729r8 for my SIP trunk, which is the only way for external calls. There's no POTS here, so, I can't test it on a PSTN.
When I tried to associate an ephone to an octo dn, I got the following:
CME(config-ephone)#button 1:7
Cannot associate conference dn 7 to button 1
01-04-2011 02:53 PM
Follow my config:
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
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol cisco
sip
registrar server expires max 3600 min 3600
localhost dns:sip.net.net
voice class codec 3
codec preference 1 ilbc
codec preference 2 g729br8
codec preference 3 g729r8
voice register global
voice-card 0
dspfarm
dsp services dspfarm
!
interface FastEthernet0/0.198
description ## Voice LAN ##
encapsulation dot1Q 198
ip address 10.80.17.1 255.255.255.0
ip nat inside
ip virtual-reassembly
sccp local FastEthernet0/0.198
sccp ccm 10.80.17.1 identifier 1 version 7.0
sccp
!
sccp ccm group 1
associate ccm 1 priority 1
associate profile 2 register confer1
associate profile 1 register xcode1
!
dspfarm profile 1 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
maximum sessions 2
associate application SCCP
!
dspfarm profile 2 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 2
associate application SCCP
!
!
dial-peer voice 5 voip
description LOCAL - TELLFREE
translation-profile outgoing PSTN_Outgoing
max-conn 5
destination-pattern 0[2-9].......
voice-class codec 3
session protocol sipv2
session target sip-server
session transport udp
dtmf-relay h245-alphanumeric rtp-nte
sip-ua
credentials username xxxxx
authentication username xxxxxxx
no remote-party-id
retry invite 2
retry register 10
timers connect 100
registrar dns:sip.net.net expires 3600
sip-server dns:sip.net.net
host-registrar
!
!
!
telephony-service
sdspfarm units 2
sdspfarm transcode sessions 2
sdspfarm tag 1 xcode1
sdspfarm tag 2 confer1
conference hardware
em logout 0:0 0:0 0:0
max-ephones 20
max-dn 20
ip source-address 10.80.17.1 port 2000
cnf-file location flash:
cnf-file perphone
load 7960-7940 P00308000500
time-zone 17
time-format 24
date-format dd-mm-yy
max-conferences 8 gain -6
call-forward pattern 8...
call-forward pattern 9T
moh music-on-hold.au
multicast moh 239.15.10.1 port 2000
transfer-system full-consult
transfer-pattern 8...
transfer-pattern 9T
secondary-dialtone 0
create cnf-files version-stamp 7960 Jan 04 2011 20:19:10
!
!
ephone-dn 2 dual-line
number 8003 secondary "siplogin" no-reg both
pickup-group 10
description User1
name User1
!
!
ephone-dn 7 dual-line
number 8007 secondary "siplogin" no-reg both
pickup-group 10
description User2
name User2
!
ephone 2
device-security-mode none
mac-address 0000.1111.2222
ephone-template 1
type 7960
keep-conference endcall
button 1:2
ephone 11
device-security-mode none
mac-address 1111.2222.3333
type 7960
keep-conference endcall
button 1:7
01-05-2011 07:10 AM
Hi,
I see that you have a hardware conference bridge in your CME. Try removing "conference hardware" under the telephony service mode.
Also add "conference ad-hoc" command under the ephone-dn participating in the conference. Try a test call.
HTH
Regards
Nitesh
PS: Pl rate helpful posts.
01-06-2011 04:14 AM
I did remove the "conference hardware" from telephony-services but when I add the "conference ad-hoc" into DN I got the following message:
CME(config-ephone-dn)#conference ad-hoc
DN 7 is tied to ephone 7 button 1
Remove dn 7 from the ephone button config
I did remove the "button 1:7" from the ephone and when I add it again I got the following message:
Cannot associate conference dn 7 to button 1
01-26-2011 05:57 AM
I got it:
By adding the following config:
ephone-dn 13 octo-line
number A100
conference ad-hoc
ephone-dn 14 octo-line
number A101
conference ad-hoc
07-13-2011 06:24 PM
Hi Leandro,
I have a same problem and still didnt get why that commands works.
Can you give me a litlle explanation on this?
07-14-2011 05:24 AM
Hello Takashi,
First....let me know a little bit more about your scenario.
If you could post your config and/or debug error messages....that would be helpful.
Do you have problems only on external calls only? what about internals?
Do you use SIP trunk?
07-17-2011 04:11 PM
Thanks for your reply.
Actually, when I did your suggested command my conference also worked in hardware. (Before it didnt worked for internal and external)
However, I still dont understand why that command allow all the phone to initiate the conference call.
I thought putting each ephone as a conference mode, but you made a dummy ephone-dn and this allowed to work.
I just want to know what this command is actually doing.
Yes, we use SIP trunk.
01-15-2014 01:29 AM
thanks all for your comments which is really helped me,
but i have a wiered problem, i have h323 trunk between my CME to CUCM , when i try to add some one registered on CUCM 8.6 it gives me "unknown number"...
but when i call someone on CUCM then add anyone on CME the conference goes fine??
would anyone kindly help me solving that issue??
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