07-16-2013 03:05 PM - edited 03-16-2019 06:23 PM
Hi,
I need when the calls come from my PSTN line number 88887777 (voice port 0/1/1), the router dial over my sip trunk to number 03399883381..
But when i do the test, the follow error happens:
003481: Jul 16 22:12:07.926: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:03399883381@X.X.X.X:5060 SIP/2.0
Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK1365BB9
From: <sip:gila.local>;tag=146C004-236C
To: <sip:03399883381@X.X.X.X>
Date: Tue, 16 Jul 2013 22:12:07 GMT
Call-ID: 96BD1769-ED9B11E2-AED7D7ED-C84F3976@gila.local
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 2528974697-3986362850-2933118957-3360635254
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIF Y, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 15
Timestamp: 1374012727
Contact: <sip:y.y.y.y:5060>
Diversion: <sip:787@y.y.y.y>;privacy=off;reason=unconditional;counter=1;s creen=no
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 310
v=0
o=CiscoSystemsSIP-GW-UserAgent 3685 6234 IN IP4 y.y.y.y
s=SIP Call
c=IN IP4 y.y.y.y
t=0 0
m=audio 17930 RTP/AVP 0 8 101 19
c=IN IP4 y.y.y.y
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
a=direction:passive
003482: Jul 16 22:12:07.938: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK1365BB9;rport=60371;received= y.y.y.y
From: <sip:gila.local>;tag=146C004-236C
To: <sip:03399883381@X.X.X.X>
Call-ID: 96BD1769-ED9B11E2-AED7D7ED-C84F3976@gila.local
CSeq: 101 INVITE
Server: nt
Content-Length: 0
003483: Jul 16 22:12:07.942: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK1365BB9;rport=60371;received= y.y.y.y
From: <sip:gila.local>;tag=146C004-236C
To: <sip:03399883381@X.X.X.X>;tag=18f01cbe78aa7d69fd8f3e0a8ea294be.a643
Call-ID: 96BD1769-ED9B11E2-AED7D7ED-C84F3976@gila.local
CSeq: 101 INVITE
Server: nt
Content-Length: 0
003484: Jul 16 22:12:07.946: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:03399883381@X.X.X.X:5060 SIP/2.0
Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK1365BB9
From: <sip:gila.local>;tag=146C004-236C
To: <sip:03399883381@X.X.X.X>;tag=18f01cbe78aa7d69fd8f3e0a8ea294be.a643
Date: Tue, 16 Jul 2013 22:12:07 GMT
Call-ID: 96BD1769-ED9B11E2-AED7D7ED-C84F3976@gila.local
Max-Forwards: 15
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
The router config is:
voice service voip
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
registrar server expires max 3600 min 3600
localhost dns:XXXX.local
no update-callerid
!
voice class codec 2
codec preference 1 g729r8
codec preference 2 g729br8
voice-port 0/1/1
supervisory disconnect dualtone mid-call
connection plar 787
caller-id enable
sip-ua
nat symmetric role passive
nat symmetric check-media-src
max-forwards 15
timers trying 1000
timers connect 1000
timers disconnect 1000
timers notify 1000
timers info 1000
sip-server ipv4:X.X.X.X:5060
telephony-service
video
no auto-reg-ephone
max-ephones 22
max-dn 88
ip source-address 10.176.10.1 port 2000
max-redirect 20
auto assign 1 to 1 type bri
calling-number initiator
service phone videoCapability 1
service dnis overlay
service dnis dir-lookup
timeouts interdigit 5
url services http://10.1.10.1/voiceview/common/login.do
url authentication http://10.1.10.1/voiceview/authentication/authenticate.do
time-zone 18
time-format 24
date-format dd-mm-yy
voicemail 300
max-conferences 8 gain -6
call-forward pattern .T
call-forward system redirecting-expanded
hunt-group logout HLog
moh music-on-hold.au
multicast moh 239.10.16.16 port 2000
dn-webedit
time-webedit
transfer-system full-consult dss
transfer-pattern 9.T
transfer-pattern .T
transfer-pattern 0.T
secondary-dialtone 0
create cnf-files version-stamp 7960 May 27 2013 20:13:10
ephone-dn 31 dual-line
number 787 no-reg primary
call-forward all 03399883381
!
dial-peer voice 1 voip
destination-pattern 03399883381
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
Solved! Go to Solution.
07-19-2013 08:45 PM
Hi Thiago,
Please confirm if the call flow is:
PSTN --- 0/1/0 --- CME --- IP phone --- Cfwd-all to
03399883381 via SIP trunk ?
++ Is the above call flow correct with CME or is there CUCM in the call flow ?
Based on the SIP debug messages, the outgoing Invite to provider has the Diversion header with 3-digit extension
!
Diversion: <>787@y.y.y.y>;privacy=off;reason=unconditional;counter=1;screen=no>
++ Seems like provider is considering this as higher precedence over Contact or From headers
( I cant confirm, since you seem to have modified the headers - Ensure there are valid DID's in 10-digit format , or the format Provider is expecting )
You can quickly try the below work-around of removing the Diversion header from the outgoing Invite to provider by applying the below sip profile under dial-peer voice 1:
!
voice class sip-profiles 1
response ANY sip-header Diversion remove
request ANY sip-header Diversion remove
!
dial-peer voice 1
voice-class sip profiles 1
Please test and let me know if this resolved the issue, if not, please do confirm the call flow and also provide the below debugs:
!
debug voip ccapi inout
debug ccsip messages
//And what is the format, provider is expecting to recieve calls ?
A quick test would be make normal outgoing call via the SIP trunk and check the headers and see what is the difference in Invite for outgoing call as supposed to call-forward all .
Regards
Prashanthi Velpula
07-19-2013 08:45 PM
Hi Thiago,
Please confirm if the call flow is:
PSTN --- 0/1/0 --- CME --- IP phone --- Cfwd-all to
03399883381 via SIP trunk ?
++ Is the above call flow correct with CME or is there CUCM in the call flow ?
Based on the SIP debug messages, the outgoing Invite to provider has the Diversion header with 3-digit extension
!
Diversion: <>787@y.y.y.y>;privacy=off;reason=unconditional;counter=1;screen=no>
++ Seems like provider is considering this as higher precedence over Contact or From headers
( I cant confirm, since you seem to have modified the headers - Ensure there are valid DID's in 10-digit format , or the format Provider is expecting )
You can quickly try the below work-around of removing the Diversion header from the outgoing Invite to provider by applying the below sip profile under dial-peer voice 1:
!
voice class sip-profiles 1
response ANY sip-header Diversion remove
request ANY sip-header Diversion remove
!
dial-peer voice 1
voice-class sip profiles 1
Please test and let me know if this resolved the issue, if not, please do confirm the call flow and also provide the below debugs:
!
debug voip ccapi inout
debug ccsip messages
//And what is the format, provider is expecting to recieve calls ?
A quick test would be make normal outgoing call via the SIP trunk and check the headers and see what is the difference in Invite for outgoing call as supposed to call-forward all .
Regards
Prashanthi Velpula
08-16-2013 01:34 PM
Sorry for the late, its worked!!
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