Problem to transfer call to Outside Destination

[Thiago Cella]

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

[Prashanthi Velpula · Accepted solution]

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

[Thiago Cella]

Sorry for the late, its worked!!