I have an existing SIP service provider, through which all our calls in and out over 60 trunks work perfectly. This ITSP does not use registration or authentication and does not require us to do any header manipulation. They have been our provider for about 5 years. Having recently dropped our ISDN backup telephony service we now have a second ITSP to provide call functionality on alternate number ranges in the event the primary ITSP should fail.
The new ITSP uses registration and authentication, so our previously very simple sip-ua config ( retry invite 3 and timers trying 350) now has many more commands in it. They also require us to employ a copylist and a sip profile to modify the inbound INVITE from them in order to make inbound calls work.
Inbound calls land on the CUBE, are picked up by the correct WANside incoming dialpeer, are matched to the correct LANside outgoing dialpeer, but only if I don't include the outbound dialpeer level sip profile command below. Calls are dropped with a 'no matching outbound dialpeer' message. If I remove the sip profile command, calls go through, but only to the trunk number as the ITSP requires header manipulation.Only the trunk registered number makes it to CUCM, no matter what number from the new range is dialled.
The relevant config is:
dial-peer voice 16 voip
description * INBOUND WANside *
session protocol sipv2
incoming called-number 44152456798.
voice-class sip copy-list 1
dtmf-relay rtp-nte
codec g711alaw
no vad
voice class sip-copylist 1
sip-header TO
dial-peer voice 17 voip
description * BACKUP OUTBOUND LANside to CUCM *
translation-profile outgoing PSTN_inbound
destination-pattern 44152456798.
session protocol sipv2
session target ipv4:10.42.26.72
voice-class sip profiles 1
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
codec g711alaw
no vad
voice class sip-profiles 1
request INVITE peer-header sip TO copy "sip:(.*)@" u01
request INVITE sip-header SIP-Req-URI modify ".*@(.*)" "INVITE sip:\u01@\1"
voice translation-profile PSTN_inbound
translate calling 4
translate called 3
voice translation-rule 3
rule 7 /^4415245\(6798.\)/ /\1/
voice translation-rule 4
rule 1 // /8/
#test voice translation-rule 4 441524567988
Matched with rule 1
Original number: 441524567988 Translated number: 8441524567988
is-sipgw02-b18#test voice translation-rule 3 441524567988
Matched with rule 7
Original number: 441524567988 Translated number: 67988
My main question then, if the translation can be seen to be working by 'test voice translation-rule 3' then how come 67988 is not passed to CUCM, and instead only 67980 from the INVITE is passed. In what way does the translation profile we're used to using with the main ITSP not work with the sip profile demanded by the new ITSP? Is the regex for the profile incorrect?
Thanks
Nathan.
Debug snippet
Received:
2426867: May 10 20:52:08.248: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
INVITE sip:441524567980@217.114.51.210:5060 SIP/2.0
Via: SIP/2.0/UDP 83.218.143.16:5060;branch=z9hG4bKsghogr20cg3hd7h1v2e1.1
Max-Forwards: 15
From: <sip:07814965714@87.224.0.9>;tag=B61DcrZmgZ08g
To: <sip:441524567988@83.218.143.22>
Call-ID: 76b48670-cf2e-1236-1c9d-5065f3f05b98
CSeq: 122654900 INVITE
Contact: <sip:07814965714@83.218.143.16:5060;transport=udp>
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: 275
X-FS-Support: update_display,send_info
Remote-Party-ID: <sip:07814965714@87.224.0.9>;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1525959014 1525959015 IN IP4 83.218.143.16
s=FreeSWITCH
c=IN IP4 83.218.143.16
t=0 0
m=audio 57594 RTP/AVP 8 0 18 101 13
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
2426934: May 10 20:52:08.252: //-1/74C2DC6BA525/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=16
2427001: May 10 20:52:08.252: //-1/xxxxxxxxxxxx/SIP/Info/critical/12288/sipSPICreateNewRawMsg: No Data to form The Raw Message
cisco-username=07814965714
----- ccCallInfo IE subfields -----
cisco-ani=07814965714
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=1
dest=441524567980
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
2427007: May 10 20:52:08.256: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_PROCEEDING cisco-username=07814965714
----- ccCallInfo IE subfields -----
cisco-ani=807814965714
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=1
dest=67980
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
