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call drops with cause code 28 while sending fax over siptrunk

Hi,

we yestarday configured router for sending fax over ip. we are able to recive the fax tone when calling to the sip channel configured for Rightfax.

but when we are sending outgoing fax it is failing with the causecode 28.

Fax server IP : 192.168.12.73

sip-server       : 10.200.7.157

Numbers alloted 2505481 to 2505489


the running config is

--------------------------------------------------------------------------------------------------------------------------------------------------------------------

dial-peer voice 180 voip

description **to FAX SERVER**

destination-pattern 250548.

session protocol sipv2

session target ipv4:192.168.12.73

codec g711alaw

fax protocol pass-through g711alaw

no vad

!

dial-peer voice 20 voip

description ***OUT-BOUND CALLS TO PSTN***

translation-profile outgoing OUT-SIP

destination-pattern .T

progress_ind progress enable 8

rtp payload-type cisco-codec-fax-ack 111

rtp payload-type nte 97

voice-class codec 1

session protocol sipv2

session target ipv4:10.200.7.157:5060

session transport udp

dtmf-relay rtp-nte

no vad

!

dial-peer voice 11 voip

description **Outgoing Call to SIP Trunk**

translation-profile outgoing OUT_FAX

destination-pattern 33.T

session protocol sipv2

session target ipv4:10.200.7.157

codec g711alaw

fax protocol pass-through g711alaw

no vad

!

!

sip-ua

disable-early-media 180

sip-server ipv4:10.200.7.157:5060

----------------------------------------------------------------------------------------------------------------------------------------------------------------------

The #debug voice ccapi inout  Result is

ASICO-RYD#

Jan 20 06:03:01.111: //263969/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_begin:

   Consume mask is not set. Relaying Digit 0 to dstCallId 0x40722

Jan 20 06:03:01.187: //263969/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_end:

   Consume mask is not set. Relaying Digit 0 to dstCallId 0x40722

Jan 20 06:03:02.343: //263976/xxxxxxxxxxxx/CCAPI/cc_api_caps_ind:

   Call Entry Is Not Found

Jan 20 06:03:02.347: //-1/E2278418BD54/CCAPI/cc_api_display_ie_subfields:

   cc_api_call_setup_ind_common:

   cisco-username=5481

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

   cisco-ani=5481

   cisco-anitype=0

   cisco-aniplan=0

   cisco-anipi=0

   cisco-anisi=0

   dest=334631294

   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

Jan 20 06:03:02.347: //-1/E2278418BD54/CCAPI/cc_api_call_setup_ind_common:

   Interface=0x46B20478, Call Info(

   Calling Number=5481,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),

   Called Number=334631294(TON=Unknown, NPI=Unknown),

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

   Incoming Dial-peer=20, Progress Indication=NULL(0), Calling IE Present=TRUE,

   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=263976

Jan 20 06:03:02.347: //-1/E2278418BD54/CCAPI/ccCheckClipClir:

   In: Calling Number=5481(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)

Jan 20 06:03:02.347: //-1/E2278418BD54/CCAPI/ccCheckClipClir:

   Out: Calling Number=5481(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)

Jan 20 06:03:02.347: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

Jan 20 06:03:02.347: :cc_get_feature_vsa malloc success

Jan 20 06:03:02.347: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

Jan 20 06:03:02.347:  cc_get_feature_vsa count is 5

Jan 20 06:03:02.347: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

Jan 20 06:03:02.347: :FEATURE_VSA attributes are: feature_name:0,feature_time:1181470208,feature_id:101595

Jan 20 06:03:02.347: //263976/E2278418BD54/CCAPI/cc_api_call_setup_ind_common:

   Set Up Event Sent;

   Call Info(Calling Number=5481(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),

   Called Number=334631294(TON=Unknown, NPI=Unknown))

Jan 20 06:03:02.351: //263976/E2278418BD54/CCAPI/cc_process_call_setup_ind:

   Event=0x469FAC70

Jan 20 06:03:02.351: //263976/E2278418BD54/CCAPI/ccCallSetContext:

   Context=0x47757BD8

Jan 20 06:03:02.351: //263976/E2278418BD54/CCAPI/cc_process_call_setup_ind:

   >>>>CCAPI handed cid 263976 with tag 20 to app "_ManagedAppProcess_Default"

Jan 20 06:03:02.351: //263976/E2278418BD54/CCAPI/ccCallProceeding:

   Progress Indication=NULL(0)

Jan 20 06:03:02.355: //263976/E2278418BD54/CCAPI/ccCallSetupRequest:

   Destination=, Calling IE Present=TRUE, Mode=0,

   Outgoing Dial-peer=11, Params=0x45B6AED4, Progress Indication=NULL(0)

Jan 20 06:03:02.355: //263976/E2278418BD54/CCAPI/ccCheckClipClir:

   In: Calling Number=5481(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)

Jan 20 06:03:02.355: //263976/E2278418BD54/CCAPI/ccCheckClipClir:

   Out: Calling Number=5481(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)

Jan 20 06:03:02.355: //263976/E2278418BD54/CCAPI/ccCallSetupRequest:

   Destination Pattern=33.T, Called Number=334631294, Digit Strip=FALSE

Jan 20 06:03:02.355: //263976/E2278418BD54/CCAPI/ccCallSetupRequest:

   Calling Number=5481(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),

   Called Number=334631294(TON=Unknown, NPI=Unknown),

   Redirect Number=, Display Info=

   Account Number=5481, Final Destination Flag=TRUE,

   Guid=E2278418-61FD-11E2-BD54-8C1DCA81D3CC, Outgoing Dial-peer=11

Jan 20 06:03:02.355: //263976/E2278418BD54/CCAPI/cc_api_display_ie_subfields:

   ccCallSetupRequest:

   cisco-username=5481

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

   cisco-ani=5481

   cisco-anitype=0

   cisco-aniplan=0

   cisco-anipi=0

   cisco-anisi=0

   dest=334631294

   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

Jan 20 06:03:02.359: //263976/E2278418BD54/CCAPI/ccIFCallSetupRequestPrivate:

   Interface=0x46B20478, Interface Type=3, Destination=, Mode=0x0,

   Call Params(Calling Number=5481,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),

   Called Number=334631294(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,

   Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=11, Call Count On=FALSE,

   Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)

Jan 20 06:03:02.359: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

Jan 20 06:03:02.359: :cc_get_feature_vsa malloc success

Jan 20 06:03:02.359: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

Jan 20 06:03:02.359:  cc_get_feature_vsa count is 6

Jan 20 06:03:02.359: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

Jan 20 06:03:02.359: :FEATURE_VSA attributes are: feature_name:0,feature_time:1181470424,feature_id:101596

Jan 20 06:03:02.359: //263977/E2278418BD54/CCAPI/ccIFCallSetupRequestPrivate:

   SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1

Jan 20 06:03:02.359: //263977/E2278418BD54/CCAPI/ccCallSetContext:

   Context=0x45B6AE84

Jan 20 06:03:02.359: //263976/E2278418BD54/CCAPI/ccSaveDialpeerTag:

   Outgoing Dial-peer=11

Jan 20 06:03:02.363: //263977/E2278418BD54/CCAPI/cc_api_call_proceeding:

   Interface=0x46B20478, Progress Indication=NULL(0)

Jan 20 06:03:02.435: //263977/E2278418BD54/CCAPI/cc_api_call_disconnected:

   Cause Value=28, Interface=0x46B20478, Call Id=263977

Jan 20 06:03:02.435: //263977/E2278418BD54/CCAPI/cc_api_call_disconnected:

   Call Entry(Responsed=TRUE, Cause Value=28, Retry Count=0)

Jan 20 06:03:02.435: //263976/xxxxxxxxxxxx/CCAPI/ccCallReleaseResources:

   release reserved xcoding resource.

Jan 20 06:03:02.435: //263977/E2278418BD54/CCAPI/ccCallSetAAA_Accounting:

   Accounting=1, Call Id=263977

Jan 20 06:03:02.435: //263977/E2278418BD54/CCAPI/ccCallDisconnect:

   Cause Value=28, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=28)

Jan 20 06:03:02.435: //263977/E2278418BD54/CCAPI/ccCallDisconnect:

   Cause Value=28, Call Entry(Responsed=TRUE, Cause Value=28)

Jan 20 06:03:02.439: //263977/E2278418BD54/CCAPI/cc_api_call_disconnect_done:

   Disposition=0, Interface=0x46B20478, Tag=0x0, Call Id=263977,

   Call Entry(Disconnect Cause=28, Voice Class Cause Code=0, Retry Count=0)

Jan 20 06:03:02.439: //263977/E2278418BD54/CCAPI/cc_api_call_disconnect_done:

   Call Disconnect Event Sent

Jan 20 06:03:02.439: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

Jan 20 06:03:02.439: :cc_free_feature_vsa freeing 466BCED0

Jan 20 06:03:02.439: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

Jan 20 06:03:02.439:  vsacount in free is 5

Jan 20 06:03:02.443: //263976/E2278418BD54/CCAPI/ccCallSetupRequest:

   Destination=, Calling IE Present=TRUE, Mode=0,

   Outgoing Dial-peer=20, Params=0x47746940, Progress Indication=NULL(0)

Jan 20 06:03:02.443: //263976/E2278418BD54/CCAPI/ccCheckClipClir:

   In: Calling Number=2505481(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)

Jan 20 06:03:02.443: //263976/E2278418BD54/CCAPI/ccCheckClipClir:

   Out: Calling Number=2505481(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)

Jan 20 06:03:02.443: //263976/E2278418BD54/CCAPI/ccCallSetupRequest:

   Destination Pattern=.T, Called Number=0334631294, Digit Strip=FALSE

Jan 20 06:03:02.443: //263976/E2278418BD54/CCAPI/ccCallSetupRequest:

   Calling Number=2505481(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),

   Called Number=0334631294(TON=Unknown, NPI=Unknown),

   Redirect Number=, Display Info=

   Account Number=5481, Final Destination Flag=TRUE,

   Guid=E2278418-61FD-11E2-BD54-8C1DCA81D3CC, Outgoing Dial-peer=20

Jan 20 06:03:02.443: //263976/E2278418BD54/CCAPI/cc_api_display_ie_subfields:

   ccCallSetupRequest:

   cisco-username=5481

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

   cisco-ani=2505481

   cisco-anitype=0

   cisco-aniplan=0

   cisco-anipi=0

   cisco-anisi=0

   dest=0334631294

   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

Jan 20 06:03:02.443: //263976/E2278418BD54/CCAPI/ccIFCallSetupRequestPrivate:

   Interface=0x46B20478, Interface Type=3, Destination=, Mode=0x0,

   Call Params(Calling Number=2505481,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),

   Called Number=0334631294(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,

   Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=20, Call Count On=FALSE,

   Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)

Jan 20 06:03:02.443: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

Jan 20 06:03:02.443: :cc_get_feature_vsa malloc success

Jan 20 06:03:02.443: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

Jan 20 06:03:02.443:  cc_get_feature_vsa count is 6

Jan 20 06:03:02.447: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

Jan 20 06:03:02.447: :FEATURE_VSA attributes are: feature_name:0,feature_time:1181470424,feature_id:101597

Jan 20 06:03:02.447: //263978/E2278418BD54/CCAPI/ccIFCallSetupRequestPrivate:

   SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1

Jan 20 06:03:02.447: //263978/E2278418BD54/CCAPI/ccCallSetContext:

   Context=0x477468F0

Jan 20 06:03:02.447: //263976/E2278418BD54/CCAPI/ccSaveDialpeerTag:

   Outgoing Dial-peer=20

Jan 20 06:03:02.451: //263978/E2278418BD54/CCAPI/cc_api_call_proceeding:

   Interface=0x46B20478, Progress Indication=NULL(0)

Jan 20 06:03:03.207: //263978/E2278418BD54/CCAPI/cc_api_call_alert:

   Interface=0x46B20478, Progress Indication=NULL(0), Signal Indication=SIGNAL RINGBACK(1)

Jan 20 06:03:03.211: //263978/E2278418BD54/CCAPI/cc_api_call_alert:

   Call Entry(Retry Count=0, Responsed=TRUE)

Jan 20 06:03:03.211: //263976/E2278418BD54/CCAPI/ccCallAlert:

   Progress Indication=NULL(0), Signal Indication=SIGNAL RINGBACK(1)

Jan 20 06:03:03.211: //263976/E2278418BD54/CCAPI/ccCallAlert:

   Call Entry(Responsed=TRUE, AlertSent=TRUE)

ASICO-RYD#

ASICO-RYD#

Jan 20 06:03:34.987: //263971/006B98E54403/CCAPI/ccGenerateToneInfo:

   Stop Tone On Digit=FALSE, Tone=Null,

   Tone Direction=Sum Network, Params=0x0, Call Id=263971

Jan 20 06:03:34.987: //263972/006B98E54403/CCAPI/cc_api_call_disconnected:

   Cause Value=16, Interface=0x46B20478, Call Id=263972

Jan 20 06:03:34.991: //263972/006B98E54403/CCAPI/cc_api_call_disconnected:

   Call Entry(Responsed=TRUE, Cause Value=16, Retry Count=0)

Jan 20 06:03:34.991: //263971/006B98E54403/CCAPI/ccConferenceDestroy:

   Conference Id=0xDCD7, Tag=0x0

Jan 20 06:03:34.991: //263971/xxxxxxxxxxxx/CCAPI/cc_api_bridge_drop_done:

   Conference Id=0xDCD7, Source Interface=0x46DC83FC, Source Call Id=263971,

   Destination Call Id=263972, Disposition=0x0, Tag=0x0

Jan 20 06:03:34.991: //263972/xxxxxxxxxxxx/CCAPI/cc_api_bridge_drop_done:

   Conference Id=0xDCD7, Source Interface=0x46B20478, Source Call Id=263972,

   Destination Call Id=263971, Disposition=0x0, Tag=0x0

Jan 20 06:03:34.991: //263971/006B98E54403/CCAPI/cc_generic_bridge_done:

   Conference Id=0xDCD7, Source Interface=0x46B20478, Source Call Id=263972,

   Destination Call Id=263971, Disposition=0x0, Tag=0x0

Jan 20 06:03:34.991: //263971/006B98E54403/CCAPI/ccCallDisconnect:

   Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)

Jan 20 06:03:34.991: //263971/006B98E54403/CCAPI/ccCallDisconnect:

   Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)

Jan 20 06:03:34.991: //263971/006B98E54403/CCAPI/cc_api_get_transfer_info:

   Transfer Number Is Null

Jan 20 06:03:34.991: //263972/006B98E54403/CCAPI/ccCallDisconnect:

   Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=16)

Jan 20 06:03:34.991: //263972/006B98E54403/CCAPI/ccCallDisconnect:

   Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)

Jan 20 06:03:34.999: //263971/006B98E54403/CCAPI/cc_api_call_disconnect_done:

   Disposition=0, Interface=0x46DC83FC, Tag=0x0, Call Id=263971,

   Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)

Jan 20 06:03:34.999: //263971/006B98E54403/CCAPI/cc_api_call_disconnect_done:

   Call Disconnect Event Sent

Jan 20 06:03:34.999: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

Jan 20 06:03:34.999: :cc_free_feature_vsa freeing 466BBB68

Jan 20 06:03:34.999: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

Jan 20 06:03:34.999:  vsacount in free is 5

Jan 20 06:03:35.003: //263972/006B98E54403/CCAPI/cc_api_call_disconnect_done:

   Disposition=0, Interface=0x46B20478, Tag=0x0, Call Id=263972,

   Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)

Jan 20 06:03:35.003: //263972/006B98E54403/CCAPI/cc_api_call_disconnect_done:

   Call Disconnect Event Sent

Jan 20 06:03:35.003: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

Jan 20 06:03:35.003: :cc_free_feature_vsa freeing 466BD8F0

Jan 20 06:03:35.003: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

Jan 20 06:03:35.003:  vsacount in free is 4

Jan 20 06:03:54.219: //263976/E2278418BD54/CCAPI/cc_api_call_disconnected:

   Cause Value=16, Interface=0x46B20478, Call Id=263976

Jan 20 06:03:54.219: //263976/E2278418BD54/CCAPI/cc_api_call_disconnected:

   Call Entry(Responsed=TRUE, Cause Value=16, Retry Count=0)

Jan 20 06:03:54.219: //263978/E2278418BD54/CCAPI/ccCallDisconnect:

   Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)

Jan 20 06:03:54.219: //263978/E2278418BD54/CCAPI/ccCallDisconnect:

   Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)

Jan 20 06:03:54.219: //263976/E2278418BD54/CCAPI/ccCallDisconnect:

   Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=16)

Jan 20 06:03:54.223: //263976/E2278418BD54/CCAPI/ccCallDisconnect:

   Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)

Jan 20 06:03:54.231: //263976/E2278418BD54/CCAPI/cc_api_call_disconnect_done:

   Disposition=0, Interface=0x46B20478, Tag=0x0, Call Id=263976,

   Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)

Jan 20 06:03:54.231: //263976/E2278418BD54/CCAPI/cc_api_call_disconnect_done:

   Call Disconnect Event Sent

Jan 20 06:03:54.231: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

Jan 20 06:03:54.231: :cc_free_feature_vsa freeing 466BCDF8

Jan 20 06:03:54.231: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

Jan 20 06:03:54.231:  vsacount in free is 3

Jan 20 06:03:54.267: //263978/E2278418BD54/CCAPI/cc_api_call_disconnect_done:

   Disposition=0, Interface=0x46B20478, Tag=0x0, Call Id=263978,

   Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)

Jan 20 06:03:54.267: //263978/E2278418BD54/CCAPI/cc_api_call_disconnect_done:

   Call Disconnect Event Sent

Jan 20 06:03:54.267: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

Jan 20 06:03:54.267: :cc_free_feature_vsa freeing 466BCED0

Jan 20 06:03:54.267: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

Jan 20 06:03:54.267:  vsacount in free is 2

Jan 20 06:03:54.775: //263970/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_begin:

   Consume mask is not set. Relaying Digit 9 to dstCallId 0x40721

Jan 20 06:03:54.815: //263970/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_end:

   Consume mask is not set. Relaying Digit 9 to dstCallId 0x40721

ASICO-RYD#

ASICO-RYD#

ASICO-RYD#

ASICO-RYD#u all

All possible debugging has been turned off

ASICO-RYD#

--------------------------------------------------------------------------------------------------------------------------------------------------------------------------

when i modified the dial-peer 20 , the fax call matching dial-peer 11 and disconnecting with cause code 28

Please can any one help.......

7 Accepted Solutions

Accepted Solutions

Hi Mohammed,

!

dial-peer voice 11 voip

description **Outgoing Call to SIP Trunk**

translation-profile outgoing OUT_FAX

destination-pattern 33.T

session protocol sipv2

session target ipv4:10.200.7.157

codec g711alaw

fax protocol pass-through g711alaw

no vad

!

As per the config you provided, it does show you have a voice translation profile configured. Looks like the running config isn't complete as it doesn't show the translation rules or profiles. Could you please share that as well? Also, what number is Telco expecting as the called party number i.e. how many digits?


--
Regards,
Harmit.

View solution in original post

Hi Mohammed,

Thank you for sharing the full config. A couple of things:

++     Looks like there is no voice translation profile by the name of OUT_FAX. Hence, that statement in dialpeer 11 wont do anything.

++     From the destination pattern configured on dialpeer 11, anything before the T would get stripped off. So it's important to understand whether the number you're calling is a local number / long distance / international number. If it's a local number, then should the number being sent to Telco be "4631294" or something else?

Please capture the following debugs for an outgoing test call:

debug voip ccapi inout

debug ccsip messages

Please mention the calling and called party numbers.


--
Regards,
Harmit.

View solution in original post

Hi Mohammed,

Thank you for the logs. Here is what I see:

SIP debugs:

Jan 20 07:54:38.339: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:334631294@10.200.7.157:5060 SIP/2.0

Via: SIP/2.0/UDP 10.66.200.214:5060;branch=z9hG4bKD8728F7

Remote-Party-ID: <5481>;party=calling;screen=no;privacy=off

From: <5481>;tag=C220B960-46F

To: <334631294>

Date: Sun, 20 Jan 2013 07:54:38 GMT

Call-ID: 79475A0F-620D11E2-88568C1D-CA81D3CC@10.66.200.214

Supported: timer,resource-priority,replaces

Min-SE:  1800

Cisco-Guid: 2034559190-1645023714-2287045661-3397505996

Jan 20 07:54:38.427: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 484 Address Incomplete

Via: SIP/2.0/UDP 10.66.200.214:5060;branch=z9hG4bKD8728F7

Record-Route: <10.200.7.157:5060>

Call-ID: 79475A0F-620D11E2-88568C1D-CA81D3CC@10.66.200.214

From: <5481>;tag=C220B960-46F

To: <334631294>;tag=sbc0803uafo7csf

CSeq: 101 INVITE

Reason: Q.850;cause=28;text="address incomplete"

Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"

Content-Length: 0

Based on what you said, Telco is looking for:

calling number :  2505481

called number  :  4631294

So for the calling number, 250 needs to be prepended and for the called party number, the 33 needs to be stripped off.

To convert the called party:

voice translation-rule 5

rule 1 /^33\(.......\)/ /\1/

voice translation-profile OUT_FAX

translate called 5

See if it works. If not, lets collect the same debugs for one test call. Not sure if your Telco will allow the call to mature if the calling party is just a 4 digit number, but worth a shot.

HTH.


--
Regards,
Harmit.

View solution in original post

Hi Mohammed,

Yes, like I suspected, the ITSP is going to want you to have the complete number for the calling party as well. Here is what I see from the SIP debugs:

Jan 20 08:35:56.719: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:4631294@10.200.7.157:5060 SIP/2.0

Via: SIP/2.0/UDP 10.66.200.214:5060;branch=z9hG4bKD954180D

Remote-Party-ID: <5481>;party=calling;screen=no;privacy=off

From: <5481>;tag=C2468A8C-198D

To: <4631294>

Date: Sun, 20 Jan 2013 08:35:56 GMT

Call-ID: 3E821914-621311E2-8BCA8C1D-CA81D3CC@10.66.200.214

Supported: timer,resource-priority,replaces

Min-SE:  1800

Cisco-Guid: 1048553588-1645416930-2344979485-3397505996

Jan 20 08:35:56.787: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 484 Address Incomplete

Via: SIP/2.0/UDP 10.66.200.214:5060;branch=z9hG4bKD954180D

Record-Route: <10.200.7.157:5060>

Call-ID: 3E821914-621311E2-8BCA8C1D-CA81D3CC@10.66.200.214

From: <5481>;tag=C2468A8C-198D

To: <4631294>;tag=sbc08042ehbstau

CSeq: 101 INVITE

Reason: Q.850;cause=28;text="address incomplete"

Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"

Content-Length: 0

Notice that the called party number is the 7 digit local number, which is correct. However the calling party is 4 digit which is what ITSP must be complaining about.

Try the following:

voice translation-rule 5

rule 1 /^33\(.......\)/ /\1/

voice translation-rule 6

rule 1 /^5\(...\)/ /2505\1/

voice translation-profile OUT_FAX

translate calling 6

translate called 5

HTH.


--
Regards,
Harmit.

View solution in original post

Hi Mohammed,

Excellent! Glad I could help :-)

For the next question, are you referring to 20 incoming DIDs and you want to publish only one number 2505499 for everyone to call into for sending faxes to your Rightfax? If yes, you can have the ITSP configure this hunting on their end since we do control that on our side, its Telco's job to set this up for you.

If you meant something else, please clarify further. HTH.


--
Regards,
Harmit.

View solution in original post

Hi Mohammed,

So, you want to publish only one number (2505499) to your clients and the ITSP has provided you with a DID range from 2505481 - 99. This would mean the remaining DIDs 2505481 - 98 would not be utilized and only 2505499 would be utilized, unless the ITSP configures the hunting on their end, such that, when a client dials 2505499, ITSP hunts between the DID range available and sends the call to one of the numbers depending on the hunting algorithm. When this is done, that is when the dialpeer configuration you mentioned in your last update will come in handy i.e. having a destination-pattern 25054[8-9].

So you see, the hunting would still need to be done on ITSP end.


--
Regards,
Harmit.

View solution in original post

I'm glad to be of help :-)

For outgoing faxes to the PSTN --> no change required. All 20 DID lines can be used.

For incoming faxes from the PSTN --> 2505499 will be the Pilot Point configured on ITSP end, which will then hunt based on the hunting algorithm configured on their end itself which DID line to send the call into your CUBE. So you will receive calls on all DID lines as well.

--
Regards,
Harmit.

View solution in original post

17 Replies 17

Harmit Singh
Cisco Employee
Cisco Employee

Hi Mohammed,

From your description, it sounds like incoming calls to Rightfax are successful and only outbound calls are failing, is that correct?

From the debug output, I see the incoming leg matches dialpeer 20, while outgoing leg matches dialpeer 11.

Disconnect cause code 28 means:

Invalid number format

Typical scenarios include:

the caller is calling out using a network type number (enterprise) rather instead of Unknown or National.

28

Indicates that the called party cannot be reached because the called party number is not in a valid format or is not complete.

The called party number is 334631294 as per the debugs. I see there is a voice translation profile also on the outgoing dialpeer. Can you please post the entire running config? What number is Telco expecting?

--
Regards,
Harmit.

Hi Harmit,

Thankyou very much for the quick reply....

the running config is

interface Tunnel100

description " Tunnel RYD-JED "

bandwidth 512

ip address 10.10.0.2 255.255.255.252

tunnel source 172.31.3.18

tunnel destination 172.31.217.202

!

interface Tunnel101

description " Tunnel RYD-DAM "

bandwidth 512

ip address 10.10.0.5 255.255.255.252

tunnel source 172.31.3.18

tunnel destination 172.31.229.130

!

interface Tunnel102

description " Tunnel RYD-NAS "

bandwidth 1024

ip address 10.10.0.10 255.255.255.252

tunnel source 172.31.3.18

tunnel destination 172.31.197.18

!

interface FastEthernet0/0

description Local LAN

ip address 192.168.12.5 255.255.255.0

duplex auto

speed auto

h323-gateway voip interface

h323-gateway voip bind srcaddr 192.168.12.5

!

interface FastEthernet0/1

description " Connection to STC MPLS"

ip address 172.31.3.18 255.255.255.252

duplex full

speed 100

!

interface FastEthernet0/0/0

ip address 10.66.200.214 255.255.255.252

duplex auto

speed auto

!

router eigrp 200

redistribute ospf 10 metric 512 600 100 100 1500

network 10.10.0.0 0.0.0.3

network 10.10.0.4 0.0.0.3

network 10.10.0.8 0.0.0.3

no auto-summary

!

router ospf 10

log-adjacency-changes

redistribute eigrp 200 subnets

network 192.168.12.0 0.0.0.255 area 0

!

router bgp 65412

no synchronization

bgp log-neighbor-changes

neighbor 172.31.3.17 remote-as 65000

no auto-summary

!

ip forward-protocol nd

ip route 0.0.0.0 0.0.0.0 192.168.12.3

ip route 10.200.7.156 255.255.255.252 10.66.200.213

!

!

ip http server

ip http access-class 23

ip http authentication local

ip http secure-server

ip http timeout-policy idle 60 life 86400 requests 10000

!

access-list 23 permit 10.10.10.0 0.0.0.7

access-list 23 permit 192.168.13.0 0.0.0.255

!

!

!

control-plane

!

!

!

voice-port 0/1/0

supervisory disconnect dualtone mid-call

no battery-reversal

input gain -3

output attenuation -3

echo-cancel coverage 32

cptone BE

timeouts initial 5

timeouts interdigit 3

timeouts call-disconnect 3

timeouts ringing 5

timeouts wait-release 1

timing hookflash-out 500

timing guard-out 300

timing sup-disconnect 50

connection plar opx 5400

impedance complex2

description STC

caller-id enable

!

voice-port 0/1/1

supervisory disconnect dualtone mid-call

no battery-reversal

input gain -3

output attenuation -3

echo-cancel coverage 32

cptone BE

timeouts initial 5

timeouts interdigit 3

timeouts call-disconnect 3

timeouts ringing 5

timeouts wait-release 1

timing hookflash-out 500

timing guard-out 300

timing sup-disconnect 50

connection plar opx 5400

impedance complex2

description STC

caller-id alerting dsp-pre-allocate

!

voice-port 0/1/2

supervisory disconnect dualtone mid-call

no battery-reversal

input gain -3

output attenuation -3

echo-cancel coverage 32

cptone BE

timeouts initial 5

timeouts interdigit 3

timeouts call-disconnect 3

timeouts ringing 5

timeouts wait-release 1

timing hookflash-out 500

timing guard-out 300

timing sup-disconnect 50

connection plar opx 5400

impedance complex2

description STC

caller-id alerting dsp-pre-allocate

!

voice-port 0/1/3

supervisory disconnect dualtone mid-call

no battery-reversal

input gain -3

output attenuation -3

echo-cancel coverage 32

cptone BE

timeouts initial 5

timeouts interdigit 3

timeouts call-disconnect 3

timeouts ringing 5

timeouts wait-release 1

timing hookflash-out 500

timing guard-out 300

timing sup-disconnect 50

connection plar opx 5400

impedance complex2

description STC

caller-id alerting dsp-pre-allocate

!

voice-port 0/2/0

!

voice-port 0/2/1

!

voice-port 0/3/0

supervisory disconnect dualtone mid-call

no battery-reversal

input gain -3

output attenuation -3

echo-cancel coverage 32

cptone BE

timeouts initial 5

timeouts interdigit 3

timeouts call-disconnect 3

timeouts ringing 5

timeouts wait-release 1

timing hookflash-out 500

timing guard-out 300

timing sup-disconnect 50

connection plar opx 5400

impedance complex2

description STC

caller-id alerting dsp-pre-allocate

!

voice-port 0/3/1

supervisory disconnect dualtone mid-call

no battery-reversal

input gain -3

output attenuation -3

echo-cancel coverage 32

cptone BE

timeouts initial 5

timeouts interdigit 3

timeouts call-disconnect 3

timeouts ringing 5

timeouts wait-release 1

timing hookflash-out 500

timing guard-out 300

timing sup-disconnect 50

connection plar opx 5400

impedance complex2

description STC

caller-id alerting dsp-pre-allocate

!

voice-port 0/3/2

supervisory disconnect dualtone mid-call

no battery-reversal

input gain -3

output attenuation -3

echo-cancel coverage 32

cptone BE

timeouts initial 5

timeouts interdigit 3

timeouts call-disconnect 3

timeouts ringing 5

timeouts wait-release 1

timing hookflash-out 500

timing guard-out 300

timing sup-disconnect 50

connection plar opx 5400

impedance complex2

description STC

caller-id alerting dsp-pre-allocate

!

voice-port 0/3/3

supervisory disconnect dualtone mid-call

no battery-reversal

input gain -3

output attenuation -3

echo-cancel coverage 32

cptone BE

timeouts initial 5

timeouts interdigit 3

timeouts call-disconnect 3

timeouts ringing 5

timeouts wait-release 1

timing hookflash-out 500

timing guard-out 300

timing sup-disconnect 50

connection plar opx 5400

impedance complex2

description STC

caller-id alerting dsp-pre-allocate

!

!

!

sccp local FastEthernet0/0

sccp ccm 192.168.12.189 identifier 2 version 5.0.1

sccp ccm 192.168.12.190 identifier 1 version 5.0.1

sccp

!

sccp ccm group 10

bind interface FastEthernet0/0

associate ccm 1 priority 1

associate ccm 2 priority 2

associate profile 3 register CFB123

associate profile 2 register MTP456

associate profile 1 register XCD789

!

dspfarm profile 1 transcode

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

maximum sessions 18

associate application SCCP

!

dspfarm profile 2 mtp

codec g711ulaw

maximum sessions hardware 4

maximum sessions software 500

associate application SCCP

!

!

dial-peer voice 15 voip

description incoming From STC Server to CUCM

destination-pattern ^25054..$

progress_ind progress enable 8

voice-class codec 1

session target ipv4:192.168.12.189

dtmf-relay rtp-nte h245-signal h245-alphanumeric

no vad

!

dial-peer voice 10 voip

translation-profile incoming INCO_SIP

rtp payload-type cisco-codec-fax-ack 111

rtp payload-type nte 97

session protocol sipv2

incoming called-number ^25054..$

dtmf-relay rtp-nte h245-signal h245-alphanumeric

codec g711alaw

no vad

!

dial-peer voice 40 voip

destination-pattern ^0125054..$

voice-class codec 1

session target ipv4:192.168.12.189

dtmf-relay rtp-nte h245-signal h245-alphanumeric

no vad

!

dial-peer voice 90 pots

incoming called-number .

direct-inward-dial

!

dial-peer voice 30 voip

description To CallManager - SBWPMPUB

destination-pattern 5400

voice-class codec 1

voice-class h323 1

session target ipv4:192.168.12.190

dtmf-relay h245-alphanumeric

no vad

!

dial-peer voice 31 voip

description to Callmanager-subscriber

preference 1

voice-class codec 1

voice-class h323 1

session target ipv4:192.168.12.189

dtmf-relay h245-alphanumeric

no vad

!

dial-peer voice 180 voip

description **to FAX SERVER**

destination-pattern 250548.

session protocol sipv2

session target ipv4:192.168.12.73

codec g711alaw

fax protocol pass-through g711alaw

no vad

!

dial-peer voice 20 voip

description ***TO-STC-LOCAL***

translation-profile outgoing OUT-SIP

destination-pattern .T

progress_ind progress enable 8

rtp payload-type cisco-codec-fax-ack 111

rtp payload-type nte 97

voice-class codec 1

session protocol sipv2

session target ipv4:10.200.7.157:5060

session transport udp

dtmf-relay rtp-nte

no vad

!

dial-peer voice 11 voip

description **Outgoing Call to SIP Trunk**

translation-profile outgoing OUT_FAX

destination-pattern 33.T

session protocol sipv2

session target ipv4:10.200.7.157

codec g711alaw

fax protocol pass-through g711alaw

no vad

!

!

sip-ua

disable-early-media 180

sip-server ipv4:10.200.7.157:5060

!

banner exec ^CC

The voice translation profile is not configured.....

Hi Mohammed,

!

dial-peer voice 11 voip

description **Outgoing Call to SIP Trunk**

translation-profile outgoing OUT_FAX

destination-pattern 33.T

session protocol sipv2

session target ipv4:10.200.7.157

codec g711alaw

fax protocol pass-through g711alaw

no vad

!

As per the config you provided, it does show you have a voice translation profile configured. Looks like the running config isn't complete as it doesn't show the translation rules or profiles. Could you please share that as well? Also, what number is Telco expecting as the called party number i.e. how many digits?


--
Regards,
Harmit.

Hi Harmit,

that is the config i have..

i also got the dubt that there is no translation profile configured on router just two dialpeers for the fax 11 and 180

and for voice calls dial-peer 20

and of Telco the range is not any number like local national and international also....

for local [1-9]

and for national 0

and for international 00

can i configure the translation profile as

voice translation-rule 5

rule 1 /^\(33\)\(..............\)/ /\2/

voice translation-profile OUT_FAX

translate calling 5


i am attaching the full running config here

Hi Mohammed,

Thank you for sharing the full config. A couple of things:

++     Looks like there is no voice translation profile by the name of OUT_FAX. Hence, that statement in dialpeer 11 wont do anything.

++     From the destination pattern configured on dialpeer 11, anything before the T would get stripped off. So it's important to understand whether the number you're calling is a local number / long distance / international number. If it's a local number, then should the number being sent to Telco be "4631294" or something else?

Please capture the following debugs for an outgoing test call:

debug voip ccapi inout

debug ccsip messages

Please mention the calling and called party numbers.


--
Regards,
Harmit.

Hi Harmit,

when we are sending outgoing fax the prefix digit should be 33 and must be stripped at the gateway and send the actual number , in this case

the calling number :  2505481

      called number :  4631294(local)

an the person sending fax sould prefix 33 to any number he sends fax,

I am attaching the debug files here

It is production network so the debug will be included with many voice calls also.

it might be annoying for you, sorry for this

Hi Mohammed,

Thank you for the logs. Here is what I see:

SIP debugs:

Jan 20 07:54:38.339: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:334631294@10.200.7.157:5060 SIP/2.0

Via: SIP/2.0/UDP 10.66.200.214:5060;branch=z9hG4bKD8728F7

Remote-Party-ID: <5481>;party=calling;screen=no;privacy=off

From: <5481>;tag=C220B960-46F

To: <334631294>

Date: Sun, 20 Jan 2013 07:54:38 GMT

Call-ID: 79475A0F-620D11E2-88568C1D-CA81D3CC@10.66.200.214

Supported: timer,resource-priority,replaces

Min-SE:  1800

Cisco-Guid: 2034559190-1645023714-2287045661-3397505996

Jan 20 07:54:38.427: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 484 Address Incomplete

Via: SIP/2.0/UDP 10.66.200.214:5060;branch=z9hG4bKD8728F7

Record-Route: <10.200.7.157:5060>

Call-ID: 79475A0F-620D11E2-88568C1D-CA81D3CC@10.66.200.214

From: <5481>;tag=C220B960-46F

To: <334631294>;tag=sbc0803uafo7csf

CSeq: 101 INVITE

Reason: Q.850;cause=28;text="address incomplete"

Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"

Content-Length: 0

Based on what you said, Telco is looking for:

calling number :  2505481

called number  :  4631294

So for the calling number, 250 needs to be prepended and for the called party number, the 33 needs to be stripped off.

To convert the called party:

voice translation-rule 5

rule 1 /^33\(.......\)/ /\1/

voice translation-profile OUT_FAX

translate called 5

See if it works. If not, lets collect the same debugs for one test call. Not sure if your Telco will allow the call to mature if the calling party is just a 4 digit number, but worth a shot.

HTH.


--
Regards,
Harmit.

dear harmit,

thaks a ton for your efforts to solve the issue...

should we define another translation-profile for the calling number to be converted to the full 2505481

the debugs i callected after configuring translation rule are

Hi Mohammed,

Yes, like I suspected, the ITSP is going to want you to have the complete number for the calling party as well. Here is what I see from the SIP debugs:

Jan 20 08:35:56.719: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:4631294@10.200.7.157:5060 SIP/2.0

Via: SIP/2.0/UDP 10.66.200.214:5060;branch=z9hG4bKD954180D

Remote-Party-ID: <5481>;party=calling;screen=no;privacy=off

From: <5481>;tag=C2468A8C-198D

To: <4631294>

Date: Sun, 20 Jan 2013 08:35:56 GMT

Call-ID: 3E821914-621311E2-8BCA8C1D-CA81D3CC@10.66.200.214

Supported: timer,resource-priority,replaces

Min-SE:  1800

Cisco-Guid: 1048553588-1645416930-2344979485-3397505996

Jan 20 08:35:56.787: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 484 Address Incomplete

Via: SIP/2.0/UDP 10.66.200.214:5060;branch=z9hG4bKD954180D

Record-Route: <10.200.7.157:5060>

Call-ID: 3E821914-621311E2-8BCA8C1D-CA81D3CC@10.66.200.214

From: <5481>;tag=C2468A8C-198D

To: <4631294>;tag=sbc08042ehbstau

CSeq: 101 INVITE

Reason: Q.850;cause=28;text="address incomplete"

Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"

Content-Length: 0

Notice that the called party number is the 7 digit local number, which is correct. However the calling party is 4 digit which is what ITSP must be complaining about.

Try the following:

voice translation-rule 5

rule 1 /^33\(.......\)/ /\1/

voice translation-rule 6

rule 1 /^5\(...\)/ /2505\1/

voice translation-profile OUT_FAX

translate calling 6

translate called 5

HTH.


--
Regards,
Harmit.

Dear harmit,

hurray.........

thanks a lot.....really you are amazing, its working fine.

one more question

i want to use the 20 lines 2505481 to 2505499

i want to create hunting on those lines and the pilot number should be 2505499

how can i do that....

once again thanks a lot bro......

Hi Mohammed,

Excellent! Glad I could help :-)

For the next question, are you referring to 20 incoming DIDs and you want to publish only one number 2505499 for everyone to call into for sending faxes to your Rightfax? If yes, you can have the ITSP configure this hunting on their end since we do control that on our side, its Telco's job to set this up for you.

If you meant something else, please clarify further. HTH.


--
Regards,
Harmit.

hi harmit,

yes, i want to publish only one number(2505499) to which the faxes should be sent.

can' t i do that from my side....

at present the dial peer is for 10 lines 250548.

if i want to add another 10 lines 250549. then do i have to configure another dial-peer

dial-peer voice 182 voip

description **to FAX SERVER**

destination-pattern 250549.     or    25054[8-9]. in dial-peer 180 only

session protocol sipv2

session target ipv4:192.168.12.73

codec g711alaw

fax protocol pass-through g711alaw

no vad

Hi Mohammed,

So, you want to publish only one number (2505499) to your clients and the ITSP has provided you with a DID range from 2505481 - 99. This would mean the remaining DIDs 2505481 - 98 would not be utilized and only 2505499 would be utilized, unless the ITSP configures the hunting on their end, such that, when a client dials 2505499, ITSP hunts between the DID range available and sends the call to one of the numbers depending on the hunting algorithm. When this is done, that is when the dialpeer configuration you mentioned in your last update will come in handy i.e. having a destination-pattern 25054[8-9].

So you see, the hunting would still need to be done on ITSP end.


--
Regards,
Harmit.

Thank you very much Harmit,

i will ask The ITSP for the Hunting,at present i can send the faxes from the 20 lines but receiving will be from only one number which i publish.right?