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VGW Sip trunk Invite looks to field "to" in header for calling party

Hello Community, the sip provider send us the authentication name in sip invite header instead of the called party. The called party in the field "to" is right. I try to use sip profiles to manipulate the voice gateway that they look into the "to" field for the calling party, but I it works not.

Have I any chance to manipulate the voice gateway that they looks into the "to" field?

Any help are appreciate.

Thanks Armin

549051: Dec 18 18:27:54.702:  //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Calling  Number=D2000101000, Called  Number=D2000101000, Peer Info  Type=DIALPEER_INFO_SPEECH

HTH, please rate all useful posts!       

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1 ACCEPTED SOLUTION

Accepted Solutions
VIP Super Bronze

VGW Sip trunk Invite looks to field "to" in header for calling p

NB: this is an update on my last post..I deleted it

I can see that you are doing early offer on your CUBE. Do you have inbound and outbound fast start enabled on CUCM? You need to enable fast start for the H323 gateway on CUCM. You are going to need an MTP for this, so ensure you have one..Then do another test call and send me the ff logs..

debug voip ccapi inout

debug h225 asn1

debug h245 asn1

debug ccsip messages

Please use this link to learn how to properly collect logs. Attach the logs here, dont paste them..

https://supportforums.cisco.com/docs/DOC-16310

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
17 REPLIES
VIP Super Bronze

VGW Sip trunk Invite looks to field "to" in header for calling p

Try this..

voice class sip-profiles 1
request INVITE sip-header TO copy "<>" u01
request INVITE sip-header SIP-Req-URI modify “.*@(.*)” “INVITE sip:\u01@213.777.22.11:5060 SIP/2.0

Then you need to add this profile to your inbound dial-peer...

Test again and send us debug ccsip messages

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
VIP Super Bronze

VGW Sip trunk Invite looks to field "to" in header for calling p

You can also try this if that doesnt work..

voice class sip-profiles 1
request INVITE peer-header TO copy "<>" u01
request INVITE sip-header SIP-Req-URI modify “.*@(.*)” “INVITE sip:\u01@213.777.22.11:5060 SIP/2.0

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

VGW Sip trunk Invite looks to field "to" in header for calling p

Hi, thanks for your help.

request INVITE sip-header TO copy "<>" u01

request INVITE sip-header SIP-Req-URI modify “.*@(.*)” “INVITE sip:\u01@213.777.22.11:5060 SIP/2.0

also not work because I have no "copy" command only modify.

HTH, please rate all useful posts!

HTH, please rate all useful posts and right answers.
VIP Super Bronze

VGW Sip trunk Invite looks to field "to" in header for calling p

You need Ios 15.1(3)T and above for you to use this feature..I suggest you upgrade your CUBE IOS

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

VGW Sip trunk Invite looks to field "to" in header for calling p

Really Really Thanks to you!!!!! We have no Cube it is a normal Router.

I will upghrade an come back to you.

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New Member

Re: VGW Sip trunk Invite looks to field "to" in header for calli

Hi, good news. The Provider change the header! We get now the right called Party in URI and the phone at CUCM rings when we get a inbound call   !!!!!!!!!

But step, by step.

The phone rings and when I pick up the Phone the calling party have still ring tone and wen have no connection. I hang up the the called party and the calling party also have still ringtone and the called party also rings again. In H323 Trace it seems when the Phone hang up the VGW get this Information. But why VGW not send 200 OK when I pick up the phone to confirm the inbound invite from Provider?

Any help are appriciate. The Trace from our Router point of view is attached.

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tkvgw: Dec 19 14:53:41.761: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:+49XXXXXXXXXXX@333.222.11.93:5060 SIP/2.0

Record-Route: <213.218.12.2>

Record-Route: <213.218.28.100>

Via: SIP/2.0/UDP 213.218.12.2;branch=z9hG4bK39c3.7ffb4b12.0

Via: SIP/2.0/UDP 213.218.28.100;branch=z9hG4bK39c3.9587b075.0

Via: SIP/2.0/UDP 213.218.28.164:5060;branch=z9hG4bK00E0F5100556C36E9D6F9804202F

From: "+4911111111111" <>+4911111111111@domain-voice.com;user=phone>;tag=00E0F5100556C36E9D6FC6403C09

To: <>+49XXXXXXXXXXX@domain-voice.com>

Call-ID: e410e00004f5-52b2fa64-67d11ce3-1ee00d80-2e7b6ac@127.0.0.1

CSeq: 33547 INVITE

Contact: <>

Max-Forwards: 68

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

Content-Type: application/sdp

Supported: 100rel, timer, replaces

P-Asserted-Identity: <>+4911111111111@domain-voice.com>

Remote-Party-ID: <>+4911111111111@domain-voice.com>;party=calling;screen=no;privacy=off

Content-Length: 441

v=0

o=- 2933601919 0 IN IP4 213.218.12.2

s=session

c=IN IP4 213.218.12.2

t=0 0

m=audio 10962 RTP/AVP 8 0 18 2 96

c=IN IP4 213.218.12.2

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=rtpmap:2 G726-32/8000

a=sendrecv

a=rtpmap:96 telephone-event/8000

m=image 10964 udptl t38

c=IN IP4 213.218.12.2

a=T38FaxVersion:0

a=T38MaxBitRate:14400

a=T38FaxUdpEC:t38UDPRedundancy

a=T38FaxRateManagement:transferredTCF

552557: Dec 19 14:53:41.789: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 213.218.12.2;branch=z9hG4bK39c3.7ffb4b12.0,SIP/2.0/UDP 213.218.28.100;branch=z9hG4bK39c3.9587b075.0,SIP/2.0/UDP 213.218.28.164:5060;branch=z9hG4bK00E0F5100556C36E9D6F9804202F

From: "+4911111111111" <>+4911111111111@domain-voice.com;user=phone>;tag=00E0F5100556C36E9D6FC6403C09

To: <>+49XXXXXXXXXXX@domain-voice.com>

Date: Thu, 19 Dec 2013 13:53:41 GMT

Call-ID: e410e00004f5-52b2fa64-67d11ce3-1ee00d80-2e7b6ac@127.0.0.1

CSeq: 33547 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

552558: Dec 19 14:53:41.809: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 213.218.12.2;branch=z9hG4bK39c3.7ffb4b12.0,SIP/2.0/UDP 213.218.28.100;branch=z9hG4bK39c3.9587b075.0,SIP/2.0/UDP 213.218.28.164:5060;branch=z9hG4bK00E0F5100556C36E9D6F9804202F

From: "+4911111111111" <>+4911111111111@domain-voice.com;user=phone>;tag=00E0F5100556C36E9D6FC6403C09

To: <>+49XXXXXXXXXXX@domain-voice.com>;tag=FA6FA0C4-26B8

Date: Thu, 19 Dec 2013 13:53:41 GMT

Call-ID: e410e00004f5-52b2fa64-67d11ce3-1ee00d80-2e7b6ac@127.0.0.1

CSeq: 33547 INVITE

Require: 100rel

RSeq: 6869

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

Allow-Events: telephone-event

Remote-Party-ID: <27>;party=called;screen=no;privacy=off

Contact: <>

Record-Route: <213.218.12.2>,<213.218.28.100>

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

552559: Dec 19 14:53:41.825: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

PRACK sip:+49XXXXXXXXXXX@192.168.1.233:5060 SIP/2.0

Via: SIP/2.0/UDP 213.218.12.2;branch=z9hG4bK1ac3.c58f4862.0

Via: SIP/2.0/UDP 213.218.28.100;branch=z9hG4bK1ac3.f3211d34.0

Via: SIP/2.0/UDP 213.218.28.164:5060;branch=z9hG4bK00E0F5100556C36E9D7E1CC3731F

From: "+4911111111111" <>+4911111111111@domain-voice.com;user=phone>;tag=00E0F5100556C36E9D6FC6403C09

To: <>+49XXXXXXXXXXX@domain-voice.com>;tag=FA6FA0C4-26B8

Call-ID: e410e00004f5-52b2fa64-67d11ce3-1ee00d80-2e7b6ac@127.0.0.1

CSeq: 33548 PRACK

Contact: <>

Max-Forwards: 68

RAck: 6869 33547 INVITE

Content-Length:     0

tkvgw#

552560: Dec 19 14:53:41.837: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 213.218.12.2;branch=z9hG4bK1ac3.c58f4862.0,SIP/2.0/UDP 213.218.28.100;branch=z9hG4bK1ac3.f3211d34.0,SIP/2.0/UDP 213.218.28.164:5060;branch=z9hG4bK00E0F5100556C36E9D7E1CC3731F

From: "+4911111111111" <>+4911111111111@domain-voice.com;user=phone>;tag=00E0F5100556C36E9D6FC6403C09

To: <>+49XXXXXXXXXXX@domain-voice.com>;tag=FA6FA0C4-26B8

Date: Thu, 19 Dec 2013 13:53:41 GMT

Call-ID: e410e00004f5-52b2fa64-67d11ce3-1ee00d80-2e7b6ac@127.0.0.1

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 33548 PRACK

Content-Length: 0

tkvgw#

552561: Dec 19 14:53:45.313: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

...

tkvgw#

552562: Dec 19 14:53:55.761: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 213.218.12.2;branch=z9hG4bK39c3.7ffb4b12.0,SIP/2.0/UDP 213.218.28.100;branch=z9hG4bK39c3.9587b075.0,SIP/2.0/UDP 213.218.28.164:5060;branch=z9hG4bK00E0F5100556C36E9D6F9804202F

From: "+4911111111111" <>+4911111111111@domain-voice.com;user=phone>;tag=00E0F5100556C36E9D6FC6403C09

To: <>+49XXXXXXXXXXX@domain-voice.com>;tag=FA6FA0C4-26B8

Date: Thu, 19 Dec 2013 13:53:41 GMT

Call-ID: e410e00004f5-52b2fa64-67d11ce3-1ee00d80-2e7b6ac@127.0.0.1

CSeq: 33547 INVITE

Require: 100rel

RSeq: 6870

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

Allow-Events: telephone-event

Remote-Party-ID: <27>;party=called;screen=no;privacy=off

Contact: <>

Record-Route: <213.218.12.2>,<213.218.28.100>

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

552563: Dec 19 14:53:55.777: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

PRACK sip:+49XXXXXXXXXXX@192.168.1.233:5060 SIP/2.0

Via: SIP/2.0/UDP 213.218.12.2;branch=z9hG4bK0ac3.979e4c03.0

Via: SIP/2.0/UDP 213.218.28.100;branch=z9hG4bK0ac3.639bf9a4.0

Via: SIP/2.0/UDP 213.218.28.164:5060;branch=z9hG4bK00E0F5100556C36EA2F2D10D6935

From: "+4911111111111" <>+4911111111111@domain-voice.com;user=phone>;tag=00E0F5100556C36E9D6FC6403C09

To: <>+49XXXXXXXXXXX@domain-voice.com>;tag=FA6FA0C4-26B8

Call-ID: e410e00004f5-52b2fa64-67d11ce3-1ee00d80-2e7b6ac@127.0.0.1

CSeq: 33549 PRACK

Contact: <>

Max-Forwards: 68

RAck: 6870 33547 INVITE

Content-Length:     0

tkvgw#

552564: Dec 19 14:53:55.781: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 213.218.12.2;branch=z9hG4bK0ac3.979e4c03.0,SIP/2.0/UDP 213.218.28.100;branch=z9hG4bK0ac3.639bf9a4.0,SIP/2.0/UDP 213.218.28.164:5060;branch=z9hG4bK00E0F5100556C36EA2F2D10D6935

From: "+4911111111111" <>+4911111111111@domain-voice.com;user=phone>;tag=00E0F5100556C36E9D6FC6403C09

To: <>+49XXXXXXXXXXX@domain-voice.com>;tag=FA6FA0C4-26B8

Date: Thu, 19 Dec 2013 13:53:55 GMT

Call-ID: e410e00004f5-52b2fa64-67d11ce3-1ee00d80-2e7b6ac@127.0.0.1

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 33549 PRACK

Content-Length: 0

tkvgw#

552565: Dec 19 14:53:57.765: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

UPDATE sip:+4911111111111@213.218.28.164:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.233:5060;branch=z9hG4bK1A671B33

From: <>+49XXXXXXXXXXX@domain-voice.com>;tag=FA6FA0C4-26B8

To: "+4911111111111" <>+4911111111111@domain-voice.com;user=phone>;tag=00E0F5100556C36E9D6FC6403C09

Date: Thu, 19 Dec 2013 13:53:55 GMT

Call-ID: e410e00004f5-52b2fa64-67d11ce3-1ee00d80-2e7b6ac@127.0.0.1

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

Max-Forwards: 70

Route: <213.218.12.2>,<213.218.28.100>

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

Timestamp: 1387461237

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

CSeq: 101 UPDATE

Contact: <>

Remote-Party-ID: <4599>;party=called;screen=yes;privacy=off

Content-Length: 0

552566: Dec 19 14:53:57.789: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

tkvgw#SIP/2.0 200 Ok

Via: SIP/2.0/UDP 192.168.1.233:5060;rport=53642;received=333.222.11.93;branch=z9hG4bK1A671B33

From: <>+49XXXXXXXXXXX@domain-voice.com>;tag=FA6FA0C4-26B8

To: "+4911111111111" <>+4911111111111@domain-voice.com;user=phone>;tag=00E0F5100556C36E9D6FC6403C09

Call-ID: e410e00004f5-52b2fa64-67d11ce3-1ee00d80-2e7b6ac@127.0.0.1

CSeq: 101 UPDATE

Contact: <>

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

Content-Length:     0

tkvgw#

552567: Dec 19 14:54:00.793: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 408 Request Timeout

Via: SIP/2.0/UDP 213.218.12.2;branch=z9hG4bK39c3.7ffb4b12.0,SIP/2.0/UDP 213.218.28.100;branch=z9hG4bK39c3.9587b075.0,SIP/2.0/UDP 213.218.28.164:5060;branch=z9hG4bK00E0F5100556C36E9D6F9804202F

From: "+4911111111111" <>+4911111111111@domain-voice.com;user=phone>;tag=00E0F5100556C36E9D6FC6403C09

To: <>+49XXXXXXXXXXX@domain-voice.com>;tag=FA6FA0C4-26B8

Date: Thu, 19 Dec 2013 13:53:55 GMT

Call-ID: e410e00004f5-52b2fa64-67d11ce3-1ee00d80-2e7b6ac@127.0.0.1

CSeq: 33547 INVITE

tkvgw#

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=102

Content-Length: 0

552568: Dec 19 14:54:00.801: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:+49XXXXXXXXXXX@333.222.11.93:5060 SIP/2.0

Via: SIP/2.0/UDP 213.218.12.2;branch=z9hG4bK39c3.7ffb4b12.0

From: "+4911111111111" <>+4911111111111@domain-voice.com;user=phone>;tag=00E0F5100556C36E9D6FC6403C09

Call-ID: e410e00004f5-52b2fa64-67d11ce3-1ee00d80-2e7b6ac@127.0.0.1

To: <>+49XXXXXXXXXXX@domain-voice.com>;tag=FA6FA0C4-26B8

CSeq: 33547 ACK

Content-Length: 0

tkvgw#

tkvgw#

tkvgw#

tkvgw#

tkvgw#

tkvgw#

tkvgw#

tkvgw#

tkvgw#

tkvgw#

tkvgw#

tkvgw#

tkvgw#

tkvgw#

tkvgw#

tkvgw#

tkvgw#

tkvgw#

tkvgw#

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tkvgw#

554191: Dec 19 15:31:55.743: %ISDN-6-DISCONNECT: Interface Serial0/0/0:30  disconnected from 06926547915 , call lasted 53 seconds

tkvgw#debug cch323 all

All CCH323 call tracing is enabled

tkvgw#

554192: Dec 19 15:32:06.280: %ISDN-6-CONNECT: Interface Serial0/0/0:30 is now connected to 01624076331 N/A

tkvgw#

554193: Dec 19 15:32:08.004: //-1/xxxxxxxxxxxx/H323/cch323_indicate_rt_packet_stats: callID=204097

554194: Dec 19 15:32:08.008: //-1/xxxxxxxxxxxx/H323/cch323_do_process_statistics: Entry, call_id=204097

554195: Dec 19 15:32:08.976: //-1/xxxxxxxxxxxx/H323/cch323_ct_main: SOCK 2 Event 0x1

554196: Dec 19 15:32:08.976: //-1/xxxxxxxxxxxx/H323/cch323_gw_process_read_socket: owner_data=0x4BC759C8, len=126, msgPtr=0x4BA12FC0

554197: Dec 19 15:32:08.976: //-1/xxxxxxxxxxxx/H323/cch323_gw_process_read_socket: Received msg for H.225

554198: Dec 19 15:32:08.976: //-1/xxxxxxxxxxxx/H323/cch323_h225_receiver: Received msg of type SETUPCFM_CHOSEN

554199: Dec 19 15:32:08.976: //204156/2A2AE0F1ACD4/H323/setup_cfm_ind: ====== PI = 0

554200: Dec 19 15:32:08.976: //204156/2A2AE0F1ACD4/H323/setup_cfm_ind: Set new event H225_EV_SETUP_CFM_IND

554201: Dec 19 15:32:08.976: //204156/2A2AE0F1ACD4/H323/setup_cfm_ind: Call Manager detected

554202: Dec 19 15:32:08.976: //204156/2A2AE0F1ACD4/H323/cch323_h225_receiver: SETUPCFM_CHOSEN: src address = 192.168.1.233; dest address = 192.168.1.213

554203: Dec 19 15:32:08.976: //204156/2A2AE0F1ACD4/H323/run_h225_sm: Received event H225_EV_SETUP_CFM_IND while at state H225_REQ_FS_ALERT

554204: Dec 19 15:32:08.976: //204156/2A2AE0F1ACD4/H323/h245_address_ind: ev=4

554205: Dec 19 15:32:08.976: //204156/2A2AE0F1ACD4/H323/setup_cfm_notify: status = 48100009

554206: Dec 19 15:32:08.976: //204156/2A2AE0F1ACD4/H323/cch323_h225_set_new_state: Changing from H225_REQ_FS_ALERT state to H225_WAIT_FOR_H245 state

554207: Dec 19 15:32:08.976: //-1/xxxxxxxxxxxx/H323/cch323_iev_queue_service: Dispatch 0xB internal event to H245 IWF SM

554208: Dec 19 15:32:08.976: //204156/2A2AE0F1ACD4/H323/run_h245_iwf_sm: received IWF_EV_H245_CONNECT_REQ while at state IWF_IDLE

554209: Dec 19 15:32:08.976: //204156/2A2AE0F1ACD4/H323/cch323_send_event_to_h245_connection_sm: Changing to new event H245_CONNECT_REQ_EVENT

554210: Dec 19 15:32:08.976: //204156/2A2AE0F1ACD4/H323/cch323_h245_connection_sm: state=0, event=1, ccb=4B0B622C, listen state=0

554211: Dec 19 15:32:08.976: //204156/2A2AE0F1ACD4/H323/cch323_h245_connection_sm: H245_CONNECT: Received event H245_CONNECT_REQ_EVENT while at H245_NONE state

554212: Dec 19 15:32:08.980: //204156/2A2AE0F1ACD4/H323/cch323_h245_set_new_state: Changing from H245_NONE state to H245_WAITING state

554213: Dec 19 15:32:08.980: //204156/2A2AE0F1ACD4/H323/h245_iwf_set_new_state: changing from IWF_IDLE state to IWF_AWAIT_H245_CONN state

554214: Dec 19 15:32:08.980: //204156/2A2AE0F1ACD4/H323/h245_connect_init: connection pending

554215: Dec 19 15:32:08.980: //-1/xxxxxxxxxxxx/H323/cch323_ct_main: SOCK 2 Event 0x1

554216: Dec 19 15:32:08.980: //-1/xxxxxxxxxxxx/H323/cch323_gw_process_read_socket: owner_data=0x4BC759C8, len=50, msgPtr=0x483C7B64

554217: Dec 19 15:32:08.980: //-1/xxxxxxxxxxxx/H323/cch323_gw_process_read_socket: Received msg for H.225

554218: Dec 19 15:32:08.980: //-1/xxxxxxxxxxxx/H323/cch323_h225_receiver: Received msg of type NOTIFYIND_CHOSEN

554219: Dec 19 15:32:08.980: //204156/2A2AE0F1ACD4/H323/notify_ind: Entry

554220: Dec 19 15:32:08.980: //204156/2A2AE0F1ACD4/H323/notify_ind: Rcvd NOTIFY Notification Indicator IE = 113

554221: Dec 19 15:32:08.980: //204156/2A2AE0F1ACD4/H323/notify_ind: Rcvd NOTIFY Connected Number as IE

554222: Dec 19 15:32:08.980: //204156/2A2AE0F1ACD4/H323/notify_ind: [cnum]/[oct]/[oct3a]= [27]/[0x00]/[0x81]

554223: Dec 19 15:32:08.980: //204156/2A2AE0F1ACD4/H323/cch323_put_embedded_obj_to_ccb: ccb=0x4B0B622C, tag=14, size=144

554224: Dec 19 15:32:08.980: //204156/2A2AE0F1ACD4/H323/cch323_put_embedded_obj_to_ccb: Insertion PASSED, embedded at 0x49513980

554225: Dec 19 15:32:08.980: //204156/2A2AE0F1ACD4/H323/notify_ind: Notify data embedded, mask=0x00000005

554226: Dec 19 15:32:08.980: //204156/2A2AE0F1ACD4/H323/cch323_h225_receiver: NOTIFYIND_CHOSEN: src address = 192.168.1.233; dest address = 192.168.1.213

554227: Dec 19 15:32:08.984: //204156/2A2AE0F1ACD4/H323/run_h225_sm: Received event H225_EV_NOTIFY_IND while at state H225_WAIT_FOR_H245

554228: Dec 19 15:32:08.984: //204156/2A2AE0F1ACD4/H323/notify_msg_notify: Entry

554229: Dec 19 15:32:08.984: //204156/2A2AE0F1ACD4/H323/notify_msg_notify: Notify data found, mask=0x00000005

554230: Dec 19 15:32:08.984: //-1/xxxxxxxxxxxx/H323/cch323_ct_main: SOCK 3 Event 0x1

554231: Dec 19 15:32:08.984: //204156/2A2AE0F1ACD4/H323/cch323_h245_channel_established_ind: Using fd=3 to send msgs

554232: Dec 19 15:32:08.984: //204156/2A2AE0F1ACD4/H323/cch323_send_event_to_h245_connection_sm: Changing to new event H245_ESTABLISHED_EVENT

554233: Dec 19 15:32:08.984: //204156/2A2AE0F1ACD4/H323/cch323_h245_connection_sm: state=1, event=2, ccb=4B0B622C, listen state=0

554234: Dec 19 15:32:08.984: //204156/2A2AE0F1ACD4/H323/cch323_h245_connection_sm: H245_CONNECT: Received event H245_ESTABLISHED_EVENT while at H245_WAITING state

tkvgw#

554235: Dec 19 15:32:08.984: //204156/2A2AE0F1ACD4/H323/cch323_h245_set_new_state: Changing from H245_WAITING state to H245_CONNECTED state

554236: Dec 19 15:32:08.984: //-1/xxxxxxxxxxxx/H323/cch323_iev_queue_service: Dispatch 0x2 internal event to H245 IWF SM

554237: Dec 19 15:32:08.984: //204156/2A2AE0F1ACD4/H323/run_h245_iwf_sm: received IWF_EV_H245_CONNECTED while at state IWF_AWAIT_H245_CONN

554238: Dec 19 15:32:08.984: //204156/2A2AE0F1ACD4/H323/h245_iwf_set_new_state: changing from IWF_AWAIT_H245_CONN state to IWF_H245_CONNECTED state

554239: Dec 19 15:32:08.984: //-1/xxxxxxxxxxxx/H323/cch323_ct_main: SOCK 3 Event 0x1

tkvgw#un all

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New Member

Re: VGW Sip trunk Invite looks to field "to" in header for calli

Hi again, the Provider gave me this information, he need from me a "200_OK" as comfirmation when I pick up the Phone from CUCM. But VGW don´t sent 200OK as confirmation for the invite.

After time the VGW sent a Time out as confirmation for the Invite.

Any help are appreciate.

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VIP Super Bronze

Re: VGW Sip trunk Invite looks to field "to" in header for calli

Yes they are correct, the gateway is not sending 200 OK.. Can you describe your setup? Is this a CCME gateway connected to ITSP??

Please send your sh run

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Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

Re: VGW Sip trunk Invite looks to field "to" in header for calli

Hi, Ip Phone 7975 -- CUCM -- H323 -- VGW 2811 | -->  SIP Trunk ITSP

                                                                        | -->  E1 PSTN

Building configuration...

no voice call carrier capacity active

!

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

redirect ip2ip

fax protocol pass-through g711alaw

h323

  emptycapability

  no h225 timeout keepalive

  h225 id-passthru

  h245 passthru tcsnonstd-passthru

sip

  bind control source-interface FastEthernet0/0

  bind media source-interface FastEthernet0/0

  header-passing

  localhost dns:voice.de

   outbound-proxy dns:voice.de

  early-offer forced

!

!

!

voice class codec 1

codec preference 1 g711alaw

!

voice class h323 1

  h225 timeout tcp establish 2

  h225 timeout setup 2

  call preserve

!

!

voice class sip-profiles 76

request INVITE sip-header From modify "<>49123456789@voice.de" "<>D2000101000@voice.de"

!

voice translation-rule 1

rule 1 /^00\([1-9].*\)/ /\1/ type national national plan isdn isdn

rule 2 /^000\([1-9].*\)/ /\1/ type international international plan isdn isdn

rule 3 /^49\(111111.*\)/ /\1/ type any national plan any isdn

rule 4 /\(^011111.*\)/ /\1/ type unknown unknown plan unknown unknown

rule 5 /^45..$/ /6111111\0/ type unknown national plan unknown isdn

!

voice translation-rule 2

rule 1 /^1111/ /91/

rule 2 /^91111/ /99/

rule 3 /^911111\(.*\)/ /\1/

!

voice translation-rule 3

rule 1 /^\([1-9].*\)/ /00\1/ type national national plan isdn isdn

rule 2 /^\([1-9].*\)/ /000\1/ type international international plan isdn isdn

rule 3 /^\([1-9].*\)/ /006157\1/ type subscriber national plan isdn isdn

!

voice translation-rule 4

rule 1 /^0\([1-9].*\)/ /\1/ type unknown unknown plan unknown unknown

rule 2 /^00\([1-9].*\)/ /\1/ type unknown national plan unknown isdn

rule 3 /^000\([1-9].*\)/ /\1/ type unknown international plan unknown isdn

!

voice translation-rule 6

rule 1 /^760\([0-9].*\)/ /+49\1/

!

voice translation-rule 7

rule 1 /^\+4912345678911/ /11/

rule 2 /^\+4912345678913/ /13/

rule 3 /^\+4912345678914/ /14/

rule 4 /^\+4912345678915/ /15/

rule 5 /^\+4912345678916/ /16/

rule 6 /^\+4912345678917/ /17/

rule 7 /^\+4912345678921/ /21/

rule 8 /^\+4912345678927/ /22/

rule 9 /^\+4912345678924/ /24/

rule 10 /^\+4912345678925/ /25/

rule 11 /^\+4912345678928/ /28/

rule 12 /^\+4912345678929/ /40/

!

!

!

voice translation-profile AMT_IN

translate calling 3

translate called 2

!

voice translation-profile AMT_OUT

translate calling 1

!

voice translation-profile SIP_AMT_IN

translate called 7

!

voice translation-profile SIP_AMT_OUT

translate called 6

!

voice translation-profile called_outgoing_PSTN

translate calling 30

translate called 10

!

voice translation-profile calling_incoming_pstn

translate calling 20

translate called 40

!

!

!

!

!

!

controller E1 0/0/0

pri-group timeslots 1-31

!

!

!

!

interface FastEthernet0/0

description Trunk zu TKSW1

ip address 333.111.1111.233 255.255.255.0

duplex full

speed 100

h323-gateway voip interface

!

interface FastEthernet0/1

no ip address

shutdown

duplex auto

speed auto

!

interface Serial0/0/0:15

description Anschluss Amt fuer POTS

no ip address

encapsulation hdlc

isdn switch-type primary-net5

isdn overlap-receiving T302 3000

isdn incoming-voice voice

isdn send-alerting

isdn sending-complete

no cdp enable

!

!

!

voice-port 0/0/0:15

translation-profile incoming AMT_IN

translation-profile outgoing AMT_OUT

cptone DE

bearer-cap 3100Hz

!

!

!

dial-peer voice 10 voip

description to CUCM

translation-profile outgoing SIP_AMT_IN

preference 5

destination-pattern .T

progress_ind setup enable 3

modem passthrough nse codec g711alaw

voice-class h323 1

session target ipv4:333.222.1111.444

dtmf-relay h245-alphanumeric rtp-nte h245-signal

codec g711alaw

fax-relay ecm disable

fax nsf 000000

no vad

!

dial-peer voice 11 voip

translation-profile incoming SIP_AMT_IN

translation-profile outgoing SIP_AMT_IN

preference 1

destination-pattern .T

progress_ind setup enable 3

modem passthrough nse codec g711alaw

voice-class h323 1

session target ipv4:333.222.1111.555

dtmf-relay h245-alphanumeric rtp-nte h245-signal

codec g711alaw

fax-relay ecm disable

fax rate disable

fax nsf 000000

no vad

!

dial-peer voice 100 pots

tone ringback alert-no-PI

destination-pattern 0T

progress_ind setup enable 3

progress_ind alert enable 8

progress_ind connect enable 8

fax rate disable

direct-inward-dial

port 0/0/0:15

!

dial-peer voice 40 pots

service stcapp

fax rate disable

port 0/1/0

!

dial-peer voice 5039 pots

service mva

incoming called-number 39

direct-inward-dial

!

dial-peer voice 600 pots

destination-pattern 6T

direct-inward-dial

port 0/2/0:15

forward-digits all

!

dial-peer voice 43 pots

service stcapp

fax rate disable

port 0/1/3

!

dial-peer voice 76 voip

translation-profile outgoing SIP_AMT_OUT

destination-pattern 76T

voice-class sip profiles 76

session protocol sipv2

session target dns:voice.de

session transport udp

codec g711alaw

!

dial-peer voice 80 voip

translation-profile incoming SIP_AMT_IN

session protocol sipv2

incoming called-number .T

codec g711alaw

!

!

sip-ua

credentials username D2000101000 password 7 1111111111111111111111111111111 realm voice.de

authentication username D2000101000 password 7 11111111111111111 realm voice.de

registrar dns:voice.de expires 3600

sip-server dns:voice.de

!

!

tkvgw#

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VIP Super Bronze

VGW Sip trunk Invite looks to field "to" in header for calling p

NB: this is an update on my last post..I deleted it

I can see that you are doing early offer on your CUBE. Do you have inbound and outbound fast start enabled on CUCM? You need to enable fast start for the H323 gateway on CUCM. You are going to need an MTP for this, so ensure you have one..Then do another test call and send me the ff logs..

debug voip ccapi inout

debug h225 asn1

debug h245 asn1

debug ccsip messages

Please use this link to learn how to properly collect logs. Attach the logs here, dont paste them..

https://supportforums.cisco.com/docs/DOC-16310

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

VGW Sip trunk Invite looks to field "to" in header for calling p

Good morning, you are 100% right!!!! Meantime I found a very good documentation, the content is very helpful and describe my szenario. I set fast start in/out and MTP and it works. But I will go ahead and change CUCM - VGW to a SIP Trunk, it is the better way.

http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_configuration_example09186a00808ead0f.shtml

Many thanks for your support, one more happy customer!!!!

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VIP Super Bronze

VGW Sip trunk Invite looks to field "to" in header for calling p

Glad I could help..Please mark the thread as answered, so as to help others in the future

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

Re: VGW Sip trunk Invite looks to field "to" in header for calli

At this place a short note for Cisco. It is all the time a big, no a very big big battle to find a "pragmatic" documentation that cover the most common szenarios and not explain a command or parameter hundret of times. I mean this books over 1000 pages.... and so on....

Explain Szenario and show step by step which parameter you should set!

Cisco create so many documentation, but to less "pracmatic" documentation.

For example, why I need a CUBE, in my case I Solve the Szenario w/ only a Router w/o CUBE functionality.

How and where I could find a docu Router SIP vs. CUBE functionality in short setence? I would not read the whole docu for Voice GW and Cubes to find some differences.

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VIP Super Bronze

VGW Sip trunk Invite looks to field "to" in header for calling p

Cisco is the best in documentation and if you look very well on this forum you will find a documentation for what you are trying to achieve. I suggest you take some time to know what a CUBE is..In your scenario you need a CUBE. When you need to connect two different voip legs, you need a CUBE, you cant just use a gateway and this is well documented...

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

Re: VGW Sip trunk Invite looks to field "to" in header for calli

I am not the only one. This is only one example.

https://supportforums.cisco.com/message/1164932#1164932 A short extract: " I have downloaded some documentation from Cisco but this is already to complicated."

No, I struggle all the time arround to find the right one.

How I should search? First is, I should search in IP telephony or in Voice over IP? In which category? Which terms and definitions, all people use different terms and definitions! When I use the common search functionality I can´t find documentation only threads and so on........

Why Cisco have a Documentation Guide? Because it is to much docu. I have never saw that someone have a documentation Guide for Documentation.

How I should go about that?

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VIP Super Bronze

Re: VGW Sip trunk Invite looks to field "to" in header for calli

There are so many ways to search..You can use google, you can search here in IP Telephony forum..I have written even few examples on how to configure CUBE, and the issue you face we have dealt with it several times here on the Forum. It takes patience, determination and will to succeed to find things and learn things..

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

Re: VGW Sip trunk Invite looks to field "to" in header for calli

Hello Community, I got help from a german colleague Sorry for the dealy.

request INVITE peer-header TO copy "<>" u01

request INVITE sip-header SIP-Req-URI modify “.*@(.*)” “INVITE sip:\u01@111.111.111.1:5060 SIP/2.0

or

request INVITE sip-header TO copy "<>" u01

request INVITE sip-header SIP-Req-URI modify “.*@(.*)” “INVITE sip:\u01@111.111.111.11:5060 SIP/2.0

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