Need help with sip-header Diversion between CME and SIP trunk

[chrysostomos1980]

Hi

We have  CME connected with ISP via sip trunk

We are experiencing two issues

1) with call forwarding

2) with the snr

It seems that the from in the sip header is the originating A number which is not be recognized by the ISP  which is expecting any did from the sip trunk and not any other number in the FROM header

So we need to modify the sip-header Diversion and here i need you help

DID Range: 25558050- 25558059

Pilot: 25558050 (is the number which registered with the sip trunk)

A number: 22551256

B number: 25558051

25558051 is forwarded to  99445555

I see that we sent invite to ISP but in the FROM we have 22551256 which the ISP not recognized this number/

We have to change it to sip trunk DIDs .ALSO i dont see to send 180 ringing butr we are send directly 183..Pls advice

INVITE sip:99445555@bbtb.cyta.com.cy:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.10.4:5060;branch=z9hG4bK8D2054

From: <sip:22551256@bbtb.cyta.com.cy>;tag=6BBEB8-B17

To: <sip:99445555@bbtb.cyta.com.cy>

Trace:

///////////////////////

000383: May 15 11:45:43.723: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:25558051@192.168.10.4:5060 SIP/2.0

Max-Forwards: 8

Via: SIP/2.0/UDP 10.224.42.132:5060;branch=z9hG4bK7mjzv5772gs6mkzvq2tcfm9v5

To: "25558051 25558051" <sip:25558051@bbtb.cyta.com.cy>;cscf

From: <sip:22551256@fmc.cyta.com.cy;user=phone>;tag=h7g4Esbg_1681470739-1368618399853-

Call-ID: BW1446398531505131781641981@10.224.50.81

CSeq: 335658039 INVITE

Contact: <sip:sgc_c@10.224.42.132;transport=udp>

Record-Route: <sip:10.224.42.132;transport=udp;lr>

Min-Se: 600

P-Asserted-Identity: <sip:22551256@fmc.cyta.com.cy;user=phone>

P-Called-Party-ID: <sip:25558050@bbtb.cyta.com.cy>

Privacy: none

Session-Expires: 1800

Content-Type: application/sdp

Content-Length: 491

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

Accept: application/media_control+xml

Accept: application/sdp

Accept: application/x-broadworks-call-center+xml

Accept: multipart/mixed

v=0

o=BroadWorks 140580428 1 IN IP4 10.224.42.132

s=-

c=IN IP4 10.224.42.6

t=0 0

m=audio 29920 RTP/AVP 18 8 100

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=yes

a=rtpmap:8 PCMA/8000

a=rtpmap:100 telephone-event/8000

a=fmtp:100 0-15

a=sqn: 0

a=cdsc: 1 audio RTP/AVP 18 8 100

a=cpar: a=rtpmap:18 G729/8000

a=cpar: a=fmtp:18 annexb=yes

a=cpar: a=rtpmap:8 PCMA/8000

a=cpar: a=rtpmap:100 telephone-event/8000

a=cpar: a=fmtp:100 0-15

a=cdsc: 4 image udptl t38

a=sendrecv

a=ptime:20

000384: May 15 11:45:43.735: //224/CF32F22D8173/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 10.224.42.132:5060;branch=z9hG4bK7mjzv5772gs6mkzvq2tcfm9v5

From: <sip:22551256@fmc.cyta.com.cy;user=phone>;tag=h7g4Esbg_1681470739-1368618399853-

To: "25558051 25558051" <sip:25558051@bbtb.cyta.com.cy>;cscf

Date: Wed, 15 May 2013 11:45:43 GMT

Call-ID: BW1446398531505131781641981@10.224.50.81

CSeq: 335658039 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

000385: May 15 11:45:43.735: //224/CF32F22D8173/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 10.224.42.132:5060;branch=z9hG4bK7mjzv5772gs6mkzvq2tcfm9v5

From: <sip:22551256@fmc.cyta.com.cy;user=phone>;tag=h7g4Esbg_1681470739-1368618399853-

To: "25558051 25558051" <sip:25558051@bbtb.cyta.com.cy>;cscf;tag=6BAB34-2590

Date: Wed, 15 May 2013 11:45:43 GMT

Call-ID: BW1446398531505131781641981@10.224.50.81

CSeq: 335658039 INVITE

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

Allow-Events: telephone-event

Contact: <sip:25558051@192.168.10.4:5060>

Record-Route: <sip:10.224.42.132;transport=udp;lr>

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

000386: May 15 11:45:48.731: //226/CF32F22D8173/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:99445555@bbtb.cyta.com.cy:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.10.4:5060;branch=z9hG4bK8D2054

From: <sip:22551256@bbtb.cyta.com.cy>;tag=6BBEB8-B17

To: <sip:99445555@bbtb.cyta.com.cy>

Date: Wed, 15 May 2013 11:45:48 GMT

Call-ID: D22E7ED5-BC8B11E2-817AF6C2-2862DEAD@192.168.88.14

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

Min-SE: 1800

Cisco-Guid: 3476222509-3163230690-2171860674-0677568173

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

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

CSeq: 101 INVITE

Timestamp: 1368618348

Contact: <sip:22551256@192.168.10.4:5060>

Expires: 60

Allow-Events: telephone-event

Max-Forwards: 7

Session-Expires: 1800

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 247

v=0

o=CiscoSystemsSIP-GW-UserAgent 3903 9766 IN IP4 192.168.10.4

s=SIP Call

c=IN IP4 192.168.10.4

t=0 0

m=audio 18256 RTP/AVP 8 101

c=IN IP4 192.168.10.4

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

000387: May 15 11:45:48.835: //226/CF32F22D8173/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 192.168.10.4:5060;received=10.228.67.248;branch=z9hG4bK8D2054

To: <sip:99445555@bbtb.cyta.com.cy>;tag=h7g4Esbg_2119287777-1368618404992

From: <sip:22551256@bbtb.cyta.com.cy>;tag=6BBEB8-B17

Call-ID: D22E7ED5-BC8B11E2-817AF6C2-2862DEAD@192.168.88.14

CSeq: 101 INVITE

Contact: <sip:sgc_c@10.224.42.132;transport=udp>

Record-Route: <sip:10.224.42.132;transport=udp;lr>

P-Asserted-Identity: <sip:99445555@bbtb.cyta.com.cy;user=phone>

Privacy: none

Content-Type: application/sdp

Content-Length: 223

Session: Media

P-Early-Media: sendonly

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

v=0

o=BroadWorks 140580512 1 IN IP4 10.224.42.132

s=-

c=IN IP4 10.224.42.7

t=0 0

m=audio 58606 RTP/AVP 8 101

c=IN IP4 10.224.42.7

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

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Regards
Chrysostomos

[Ayodeji Okanlawon]

Chrys,

This is a bit difficult..The second invite does not use a diversion header..rather it uses a From header. We can only change the From header if your normal calls use a different header for CLI

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

[chrysostomos1980]

Hi mate

what is your suggestion here. What is the solution? We have to find something :)

Sent from Cisco Technical Support iPhone App

[Ayodeji Okanlawon]

Can you do a test call outbound call for normal calls..I want to see if you PAI (P-asserted-identity) or remote party id--If you do then we can configure sip profiles to change the FROM to use a valid DDI

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

[Jorge Armijo]

Hi All,

What I can see is a second INVITE from CME to ITSP, with the Originator CLID, so the ITSP will reject the call if they don't know the number.

Please try:

telephony-service

calling-number local

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/command/reference/cme_c1ht.html#wp1012060

HTH

--
Jorge Armijo

Please remember to rate helpful responses and identify helpful or correct answers.

[chrysostomos1980]

Hi

i have add the above command and the call now its coming to the mobile.But we dont have audio.The CME connected directly to the  ISP modem .Maybe they have enable NAT and this create the no two way audio/??

[Ayodeji Okanlawon]

From the previous trace I can see that you are telling your ITSP to send media on ip:192.168.10.4..Can they connect to you on this IP address?

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

[chrysostomos1980]

aok

they give us a modem for the sip trunk.We have enabled dhcp @ the interface connected to modem ISP.The GW of their modem is 192.168.10.254 and our ip is 192.168.10.4/

So they use NAT right?

[Ayodeji Okanlawon]

I am not sure but most likely. You might want to speak to them and clarify things

Please rate all useful posts

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

[chrysostomos1980]

Hi

I have asked them to disable the nat.Now we have a different ip without nating

Now when i have a normal call everything looks fine

When i use SNR or call forward then  call reach the destination number with SNR or with CF BUT we dont have audio

Pls check attached trace for normal call and call forward call

[chrysostomos1980]

Aok

Did you have a time to check the traces?

As i wrote the call to th callforward reach the destination number but we dont have audio

Regards
Chrysostomos

[HRS]

Hi, 

Do you have a solution for this issue. We are facing the same issue, Able to make inbound and outbound call but Call Forward / SNR has NO WAY AUDIO issue.

~ HRS

[Robert Shaw]

Hi,

This post is three years old, it may be best starting a new discussion with debugs and configs from your own CUBE.

The resolution may be completely different.

Thanks
Rob

[HRS]

Hi,

Call Forward started working after I enabled Media Flow-Around under Voice Service Voip.

Thanks,

Hari