SIP Transfer/Forward: Wrong calling number shown to transfer-to

[Grant Bagdasarian]

Hello,

We have a SIP trunk which is connected through a SIP proxy. Our carrier did some testing yesterday to see if everything is working as it should. There was one problem, which was when forwarding/transfering a call. The problem is that the transferee's phonenumber is shown in the SIP INVITE FROM section. This should be our phonenumber. We must fix this because it causes problem with the billing.

Example:

Transferee = 1000

Transferor = 2000 (Our company)

Transfer-to = 3000

1000 is shown instead of 2000. These are just examples, we used full E.164 numbers when testing.

Is my configuration doing a blind transfer for some reason? Because it should do a consulvative transfer.

Below my configuration. What am I doing wrong?

IOS: Version 12.4(15)T9

CME: 4.1

voice service voip

allow-connections h323 to h323

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

redirect ip2ip

sip

dial-peer voice 2001 voip

description **Outgoing Call to SIP Trunk**

destination-pattern .T

voice-class codec 100

voice-class sip dtmf-relay force rtp-nte

session protocol sipv2

session target sip-server

session transport udp

dtmf-relay rtp-nte

no vad

telephony-service

transfer-system full-consult

transfer-pattern .T

[Grant Bagdasarian]

I applied a voice translation rule. It only translates the below in the INVITE to the transfer-to.

Contact:

The FROM still remains with the transferee's phone number.

[ashok_boin]

Hi Grant,

It's working as expected.

Pls refer below SIP call flows and refer Call transfer with both types (Blind and Consultative transfer). In both cases, the transferee sends final INVITE which is why 1000 is reflecting as FROM.

http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7960g_7940g/sip/1_0/english/administration/guide/appBcf.pdf

I am not sure whether any solution available. If available, it works only when 1000 also uses the same Voice GW as 2000.

Regards...

-Ashok.

[Grant Bagdasarian]

Thank you for that document. It will come in handy in the future.

I managed to fix it using the calling-number local secondary under telephony-service.

I tried with two scenarios:

1) VXML app under a dial-peer which transfers.

2) Call foward all under an ephone-dn.

Though the weird thing is that the display name in the From header is still showing 1000 when I use the VXML, but 2000 in the <2000>. Our carrier is probably after what's in the <>.

[ashok_boin]

Hi Grant,

Very happy to know that you could able to find a solution for your requirement.

Shall I request more information about your setup for the following queries to get clarity about the solution?

1. Is 1000 and 2000 using same Voice GW?

2. The releavnt GW config

3. SIP Headers change capture if you have...

As per the Cisco doc shared with you, the final INVITE should have been sent from 1000 (as requested by 2000) in the consultative call transfer so I am not clear how "sip:2000@..." created because of "call secondary" option?

Regards...

-Ashok.

[ashok_boin]

Hi Grant,

The solution you provided is still not clear to me.

As per the given call flows (Cisco doc) above, the transferree (in your case, 1000) originates final invite for both consultative and blind transfers. Then, how did your change in the CME configuration affect the transferee behavior?

Regards...

-Ashok.

[Grant Bagdasarian]

Hello,

I'm sorry for my late reply. I've been very busy.

1000 and 2000 use different GWs, but are both ours.

Below the config of our 3825 which has the SIP Trunk:

voice service voip

allow-connections h323 to h323

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

redirect ip2ip

h323

sip

!

!

voice class codec 100

codec preference 1 g711alaw

codec preference 2 g711ulaw bytes 80

codec preference 3 g723ar53

codec preference 4 g723ar63 bytes 144

codec preference 5 g723r53

codec preference 6 g723r63 bytes 120

codec preference 7 g726r16

codec preference 8 g726r24

codec preference 9 g726r32

codec preference 10 g728

codec preference 11 g729br8

codec preference 12 g729r8 bytes 50

interface GigabitEthernet0/0

description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$$ES_LAN$$FW_INSIDE$

ip address 172.0.0.1 255.255.255.0

dial-peer voice 2000 voip

description **Incomming Call to SIP Trunk**

service voiceforwarding

voice-class codec 100

voice-class sip dtmf-relay force rtp-nte

session protocol sipv2

session target sip-server

incoming called-number xxxxxxxxxx

dtmf-relay rtp-nte

no vad

dial-peer voice 2001 voip

description **Outgoing Call to SIP Trunk**

destination-pattern .T

voice-class codec 100

voice-class sip dtmf-relay force rtp-nte

no voice-clas sip authenticate redirecting-number

session protocol sipv2

session target sip-server

session transport udp

dtmf-relay rtp-nte

no vad

sip-ua

no remote-party-id

timers connect 100

mwi-server ipv4:192.0.0.1 expires 3600 port 5060 transport udp unsolicited

registrar ipv4:172.0.0.1 expires 3600

sip-server ipv4:192.0.0.1:5060

  host-registrar

!

telephony-service

calling-number local secondary

SIP INVITE from 1000 to 2000

Received:

INVITE sip:2000@xxxxxx:5060 SIP/2.0

Max-Forwards: 68

Session-Expires: 3600;refresher=uac

Min-SE: 600

Supported: timer, 100rel

To: <2000>

From: "1000" <1000>;tag=3529902009-759089

Remote-Party-Id: <1000>;screen=yes;privacy=off

Call-ID: 1180-3529902009-759085@xxxxxxx

CSeq: 1 INVITE

Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH

Via: SIP/2.0/UDP xxxxxxxx:5060;branch=z9hG4bKde4dc8605c808cfe18a610dafdfa1e2f

Contact: <1000>

Expires: 180

Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000"

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Length: 296

SIP INVITE FROM 2000 to 3000

Sent:

INVITE sip:3000@xxxxxxx:5060 SIP/2.0

Via: SIP/2.0/UDP xxxxxxx:5060;branch=z9hG4bK250F498

From: "1000" <2000>;tag=1EB42AAC-1CF2

To: <3000>

Date: Thu, 10 Nov 2011 10:24:36 GMT

Call-ID: 5DFF26C-ABD11E1-9B4DF927-20F9F0A9@xxxxxxx

Supported: 100rel,timer,resource-priority,replaces

Min-SE:  1800

Cisco-Guid: 98562668-180163041-2605381927-553250985

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Timestamp: 1320920676

Contact: <2000>

Diversion: <2000>;privacy=off;reason=follow-me;screen=no

Expires: 180

Allow-Events: kpml, telephone-event

Max-Forwards: 67

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 307

Regards,

Grant

[ashok_boin]

Excellent. I got it.

Thanks a lot, Grant.

Regards...

-Ashok.