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Beginner

dial peer not matching

What is wrong with this Inbound Dial Peer?  It doesnt seem to match.  I have even replaced it with the exact DNIS being sent from our Freeswitch box to the gateway and it still doesnt match.  Our Freeswitch box prefixes the digits.  The freeswitch box is added to the Toll Fraud allowed list.

10.21.10.250 is the GW

10.21.10.12 is the Freeswitch

dial-peer voice 201 voip
description INBOUND from FS
session protocol sipv2
incoming called-number 3336+1112223333
codec g711ulaw

VoiceGW#
*Oct 17 11:38:01.149: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:3336+1112223333@10.21.10.250 SIP/2.0
Via: SIP/2.0/UDP 10.21.10.12:5080;rport;branch=z9hG4bKNagctFetZeByp
Max-Forwards: 69
From: "1001" <sip:FreeSWITCH@10.21.10.250>;tag=Na3U2yrHgryKr
To: <sip:3336+1112223333@10.21.10.250>
Call-ID: b328f4cb-0f04-1235-41bf-005056820b2e
CSeq: 98016871 INVITE
Contact: <sip:gw+f170dc57-cdf4-4115-933c-7c9b5d4e1fed@10.21.10.12:5080;transport=udp;gw=f170dc57-cdf4-4115-933c-7c9b5d4e1fed>
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 289
X-accountcode: 10.21.10.12
X-FS-Support: update_display,send_info
Remote-Party-ID: "1001" <sip:13053284888@10.21.10.250>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1476674500 1476674501 IN IP4 10.21.10.12
s=FreeSWITCH
c=IN IP4 10.21.10.12
t=0 0
m=audio 31370 RTP/AVP 0 8 3 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20

*Oct 17 11:38:01.153: //-1/FE40F48D842D/CCAPI/cc_api_display_ie_subfields:
cc_api_call_setup_ind_common:
cisco-username=13053284888
----- ccCallInfo IE subfields -----
cisco-ani=13053284888
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=1
dest=3336+1112223333
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

*Oct 17 11:38:01.153: //-1/FE40F48D842D/CCAPI/cc_api_call_setup_ind_common:
Interface=0x23036E14, Call Info(
Calling Number=13053284888,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=3336+1112223333(TON=Unknown, NPI=Unknown),
Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
Incoming Dial-peer=0, Progress Indication=NULL(0), Calling IE Present=TRUE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=3787
*Oct 17 11:38:01.153: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

Everyone's tags (1)
5 REPLIES 5
Cisco Employee

Hi Joli,

Hi Joli,

If you check the Ivite INVITE sip:3336+1112223333@10.21.10.250 SIP/2.0

3336 is not prefixed where as it is with and operator.

Also when you give the + in the string with Incoming-called number it is taken as attribute.

Do could you send it correct CLID.

Beginner

OK so how can I fix this.

OK so how can I fix this.  This is what Freeswitch sends to the GW.  All calls are prefixed that way. Calls are sent out of Freeswitch in E.164 Format.

I could have it generic dial-peer to match everything, but the issue is that we will have different boxes connected to the GW with different prefixes

I have changed it to 

incoming called-number 3336T

and it still doesnt work.

I ultimately need to match the dial-peer strip the 3336 and send it out to the carrier.

thanks,

Cisco Employee

Hi Joli,

Hi Joli,

I am not aware as to how your rest of the calls are working but you can create a dial-peer with incoming called number . and use translation profiles to strip the prefixes:

Rule 1 can be for 3336 and for example if for other calls you are getting 4446 then you can strip it in rule 2.

voice translation-rule 1

rule 1 /^3336\(.*\)/ /\1/

rule 2 /^4446\(.*\)/ /\1/

Voice translation profile incoming

translate called 1

Aseem

(Please rate if useful)

Beginner

This is what I have and it

This is what I have and it doesnt seem to be working.  When I do

test voice transtation-rule 21 3336*1234568 it works, but it doesnt work on the dial peer correctly.  The 3336 does not get stripped.

voice translation-rule 21
rule 1 /^3336\(.*\)/ /\1/

voice translation-profile FromHomeFS
translate called 21

dial-peer voice 201 voip
description INBOUND from FS
translation-profile incoming FromHomeFS
session protocol sipv2
incoming called-number .T
codec g711ulaw

Highlighted
Rising star

3336+  means 6 is repeated on

3336+  means 6 is repeated on or more time.

http://docwiki.cisco.com/wiki/Cisco_IOS_Voice_Troubleshooting_and_Monitoring_--_Voice_Call_Debug_Filtering_on_Cisco_Voice_Gateways#Table:_Number_Matching_Examples_Using_Wildcard_Symbols

Try Escape character 3336\+1112223333. 

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