05-23-2012 12:32 PM - edited 03-16-2019 11:18 AM
Everybody say: "you can use SIP instead SCCP but will loose many features"... but nobody said what exactly which features.
Im using 10 x 3905 SIP phones and I´m unable to use the hold and transfer buttons. I just put a post asking about this issue, but
nobody shows interest.
No one cares SIP?
05-23-2012 01:11 PM
I don't think the issue here is SIP vs. SCCP, but rather the phone model. I myself have not worked with 3905 and hope I never have to, but just like with other previous models newly released phone models may experience more bugs than others.
To provide some help here, did you upgrade the phone to latest firmware?
What firmware are you using and what version of CUCM/CME?
What are you observing when you invoke these fetures?
Have you looked through bug tool?
Chris
05-23-2012 01:22 PM
Hello Chris,
the firmware on the phones is the latest, 9.2(2). The CME is 9.0 the ios is 15.2.2T1.
When I try put a call on hold pressing the pause button, the call drops, same thing when I try transfer a call pressing the transfer button. In both cases after the call is dropped, I dial tone appears.
the config:
!
voice service voip
no ip address trusted authenticate
address-hiding
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
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface Loopback0
bind media source-interface Loopback0
registrar server expires max 1200 min 300
early-offer forced
midcall-signaling passthru
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729br8
codec preference 3 g711alaw
!
!
voice register global
mode cme
source-address 192.168.100.254 port 5060
timeouts interdigit 5
max-dn 50
max-pool 25
load 3905 CP3905.9-2-1-0
authenticate register
authenticate realm cibrealm
timezone 13
time-format 24
date-format D/M/Y
tftp-path flash:
file text
create profile sync 0002315003536705
user-locale ES load CME-locale-es_ES-Spanish-8.8.2.4.tar
!
!
voice register dn 1
number 5861
call-forward b2bua noan 5555 timeout 20
allow watch
pickup-group 1
name 5861
shared-line
no-reg
label 5861
mwi
!
voice register dn 2
number 5862
allow watch
pickup-group 1
name 5862
shared-line
no-reg
label 5862
mwi
!
!
telephony-service
sdspfarm conference mute-on 111 mute-off 222
sdspfarm units 1
conference hardware
max-ephones 20
max-dn 50
ip source-address 192.168.100.254 port 2000
calling-number initiator
cnf-file location flash:
max-conferences 1 gain -6
web admin system name admin xxxxxxxxx
dn-webedit
time-webedit
transfer-system full-consult
!
05-23-2012 01:24 PM
Does it drop when you make a call between 2 IP phones or IP phone and SIP Trunk PSTN?
05-23-2012 01:43 PM
it fail when the call is originated from PSTN. I´m not sure between 2 ip phones. I will try and will let you know.
05-24-2012 06:44 AM
The issue is with the PSTN.
I´m not in the site, Will the redirect ip2ip solve this issue?.
Thanks.
05-24-2012 09:20 AM
I would suggest you add it. Also, take a look at the config guide:
HTH,
Chris
05-29-2012 11:59 AM
still have problem with this. redirect ip2ip in the incomming dial-peer do no fix this.
I saw in the release notes of the 3905 and the config guide that I need setup a template for hold, transfer and ...(other that in the moment dont remember). may be it is the key?.
may something related with supplementary-service h450 fix the transfer problem?.
Im close to give up.. I need some help.
Thanks.
09-12-2012 08:01 AM
It seems like codec mismach.
Try to use codec on 3905 directly or configure transcode resources.
Part of our working config:
voice register pool 3
id mac AAAA.BBBB.CCCC
type 3905
number 1 dn 3
cor incoming otherlist default
username u3 password 3
codec g711ulaw
voice-card 0
dspfarm
telephony-service
sdspfarm transcode sessions 4
transfer-pattern .T
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmetrnsc.html
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/srnd/design/guide/clproc.html
HTH
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