01-19-2013 10:18 PM - edited 03-16-2019 03:15 PM
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.......
Solved! Go to Solution.
01-19-2013 11:04 PM
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.
01-19-2013 11:42 PM
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.
01-20-2013 12:28 AM
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=off5481>
From: <5481>;tag=C220B960-46F5481>
To: <334631294>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>10.200.7.157:5060>
Call-ID: 79475A0F-620D11E2-88568C1D-CA81D3CC@10.66.200.214
From: <5481>;tag=C220B960-46F5481>
To: <334631294>;tag=sbc0803uafo7csf334631294>
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.
01-20-2013 01:10 AM
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=off5481>
From: <5481>;tag=C2468A8C-198D
5481>
To: <4631294>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>10.200.7.157:5060>
Call-ID: 3E821914-621311E2-8BCA8C1D-CA81D3CC@10.66.200.214
From: <5481>;tag=C2468A8C-198D5481>
To: <4631294>;tag=sbc08042ehbstau4631294>
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.
01-20-2013 01:35 AM
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.
01-20-2013 02:08 AM
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.
01-20-2013 02:25 AM
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.
01-19-2013 10:37 PM
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:
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.
01-19-2013 10:54 PM
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.....
01-19-2013 11:04 PM
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.
01-19-2013 11:19 PM
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
01-19-2013 11:42 PM
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.
01-20-2013 12:07 AM
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
01-20-2013 12:28 AM
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=off5481>
From: <5481>;tag=C220B960-46F5481>
To: <334631294>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>10.200.7.157:5060>
Call-ID: 79475A0F-620D11E2-88568C1D-CA81D3CC@10.66.200.214
From: <5481>;tag=C220B960-46F5481>
To: <334631294>;tag=sbc0803uafo7csf334631294>
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.
01-20-2013 12:47 AM
01-20-2013 01:10 AM
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=off5481>
From: <5481>;tag=C2468A8C-198D
5481>
To: <4631294>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>10.200.7.157:5060>
Call-ID: 3E821914-621311E2-8BCA8C1D-CA81D3CC@10.66.200.214
From: <5481>;tag=C2468A8C-198D5481>
To: <4631294>;tag=sbc08042ehbstau4631294>
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.
01-20-2013 01:24 AM
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......
01-20-2013 01:35 AM
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.
01-20-2013 01:58 AM
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
01-20-2013 02:08 AM
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.
01-20-2013 02:14 AM
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?
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