Hi, in the voice register pool 1 we have set the codec g711alaw. Do we change it?

The DNs of the SIP phones are 2.., how do we configure the incoming dial-peer?

Thank you very much.

Regards.

Hi, please use g711u not a. Incoming called-number with a pattern that covers all the sip phones, like "." ot "70."

Hi, I have done a debug ("debug voice ccapi inout") when the SIP phone in site A calls to SCCP phone in site B, and it returns:

48.535: //2266/xxxxxxxxxxxx/CCAPI/cc_api_caps_ind:

Call Entry Is Not Found

035950: *Apr 14 15:19:48.535: //-1/0E32B8E59D97/CCAPI/cc_api_display_ie_subfield

s:

cc_api_call_setup_ind_common:

cisco-username=03219

----- ccCallInfo IE subfields -----

cisco-ani=03219

cisco-anitype=0

cisco-aniplan=0

cisco-anipi=0

cisco-anisi=0

dest=001404

cisco-desttype=0

cisco-destplan=0

cisco-rdie=FFFFFFFF

cisco-rdn=

cisco-rdntype=0

cisco-rdnplan=0

cisco-rdnpi=-1

cisco-rdnsi=-1

cisco-redirectreason=-1 fwd_final_type =0

final_redirectNumber =

hunt_group_timeout =0

035951: *Apr 14 15:19:48.535: //-1/0E32B8E59D97/CCAPI/cc_api_call_setup_ind_comm

on:

Interface=0x66CAF64C, Call Info(

Calling Number=03219,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not

Screened, Presentation=Allowed),

Called Number=001404(TON=Unknown, NPI=Unknown),

Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=T

RUE,

Incoming Dial-peer=40003, Progress Indication=NULL(0), Calling IE Present=TRU

E,

Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALS

E), Call Id=2266

035952: *Apr 14 15:19:48.539: //-1/0E32B8E59D97/CCAPI/ccCheckClipClir:

In: Calling Number=03219(TON=Unknown, NPI=Unknown, Screening=Not Screened, Pr

esentation=Allowed)

035953: *Apr 14 15:19:48.539: //-1/0E32B8E59D97/CCAPI/ccCheckClipClir:

Out: Calling Number=03219(TON=Unknown, NPI=Unknown, Screening=Not Screened, P

resentation=Allowed)

035954: *Apr 14 15:19:48.539: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

035955: *Apr 14 15:19:48.539: :cc_get_feature_vsa malloc success

035956: *Apr 14 15:19:48.539: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

035957: *Apr 14 15:19:48.539: cc_get_feature_vsa count is 1

035958: *Apr 14 15:19:48.539: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

035959: *Apr 14 15:19:48.539: :FEATURE_VSA attributes are: feature_name:0,feartu

re_time:1726702304,feature_id:2317

035960: *Apr 14 15:19:48.539: //2266/0E32B8E59D97/CCAPI/cc_api_call_setup_ind_co

mmon:

Set Up Event Sent;

Call Info(Calling Number=03219(TON=Unknown, NPI=Unknown, Screening=Not Screen

ed, Presentation=Allowed),

Called Number=001404(TON=Unknown, NPI=Unknown))

035961: *Apr 14 15:19:48.539: //2266/0E32B8E59D97/CCAPI/cc_process_call_setup_in

d:

Event=0x66EC4530

035962: *Apr 14 15:19:48.543: //2266/0E32B8E59D97/CCAPI/ccCallSetContext:

Context=0x67956B4C

035963: *Apr 14 15:19:48.543: //2266/0E32B8E59D97/CCAPI/cc_process_call_setup_in

d:

>>>>CCAPI handed cid 2266 with tag 40003 to app "_ManagedAppProcess_Default"

035964: *Apr 14 15:19:48.543: //2266/0E32B8E59D97/CCAPI/ccCallProceeding:

Progress Indication=NULL(0)

035965: *Apr 14 15:19:48.547: //-1/xxxxxxxxxxxx/CCAPI/cc_update_feature_vsa:

035966: *Apr 14 15:19:48.547: : updating existing feature vsa

035967: *Apr 14 15:19:48.547: //-1/xxxxxxxxxxxx/CCAPI/cc_update_feature_vsa:

035968: *Apr 14 15:19:48.547: feature call basic

035969: *Apr 14 15:19:48.547: //2266/0E32B8E59D97/CCAPI/ccCallDisconnect:

Cause Value=3, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Ca

use=0)

035970: *Apr 14 15:19:48.547: //2266/0E32B8E59D97/CCAPI/ccCallDisconnect:

Cause Value=3, Call Entry(Responsed=TRUE, Cause Value=3)

035971: *Apr 14 15:19:48.727: //2266/0E32B8E59D97/CCAPI/cc_api_call_disconnect_d

one:

Disposition=0, Interface=0x66CAF64C, Tag=0x0, Call Id=2266,

Call Entry(Disconnect Cause=3, Voice Class Cause Code=0, Retry Count=0)

The cause value = 3 in CallDisconnect means "no route to destination". But there are a dial-peer voip with destination-pattern 001..., and session target with de ip address of the remote site. Why the SCCP phone match the dial-peer and not the SIP phone?

Thanks.

Regards.

I think the router at site A doesn`t match the corresponding Dial Peer because when a SCCP phone dials the same number the call is established ok.

Thanks for the insight, I'll keep you posted.

Regards,

Ok everything is working fine with a SIP Dial Peer and g711a (why g711u? In Spain we use g711a).

Is there any possible way to make it work with g729? I've changed the Dial Peer codec and Voice Register Pool and I get this error "cause value=65" which means "bearer capabilities not implemented".

I'm experimenting still, Keep you posted!

Regards,

Hi, g711a is only used on digital interface. On voip, uncompressed is always g711u.

You get bearer capability problems when you try to to use compressed voice and one endpoint cannot. It can be complicated to use as you might need transconding, so begin with g711u first.