Unanswered Question
May 23rd, 2012
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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?

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Chris Deren Wed, 05/23/2012 - 13:11
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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?


Rafael Jimenez Wed, 05/23/2012 - 13:22
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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


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


  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 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-



voice register dn  1

number 5861

call-forward b2bua noan 5555 timeout 20

allow watch

pickup-group 1

name 5861



label 5861



voice register dn  2

number 5862

allow watch

pickup-group 1

name 5862



label 5862





sdspfarm conference mute-on 111 mute-off 222

sdspfarm units 1

conference hardware

max-ephones 20

max-dn 50

ip source-address port 2000

calling-number initiator

cnf-file location flash:

max-conferences 1 gain -6

web admin system name admin xxxxxxxxx



transfer-system full-consult


Chris Deren Wed, 05/23/2012 - 13:24
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Does it drop when you make a call between 2 IP phones or IP phone and SIP Trunk PSTN?

Rafael Jimenez Wed, 05/23/2012 - 13:43
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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.

Rafael Jimenez Thu, 05/24/2012 - 06:44
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The issue is with the PSTN.

I´m not in the site, Will the redirect ip2ip solve this issue?.


Rafael Jimenez Tue, 05/29/2012 - 11:59
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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.


Eugene Zuevski Wed, 09/12/2012 - 08:01
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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


type 3905

number 1 dn 3

cor incoming otherlist default

username u3 password 3

codec g711ulaw

voice-card 0



sdspfarm transcode sessions 4

transfer-pattern .T



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