cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
1564
Views
10
Helpful
10
Replies

Ringing Sound for the other party

tiger_401
Level 1
Level 1

Dear All,

I am running Call Manager 8.5.

My question is that when i pick up the hand set of the IP Phone and dials any number outside i does not hear the ringing voice in my ip phone.

It just connects the call without any ringing sound.

Like for example when we dial from our cell to any other number we hear the ringing sound but once i dial from my ip phone i don’t hear this sound.

Can you please tell me how i can enable this option in the Call Manger i would be very thankful to you.

Regards,

Malik.

2 Accepted Solutions

Accepted Solutions

Hi - First of all, I think its generaly a good idea to get rid of the h323 leg and replace it with SIP. All you would need to do is configure SIP trunk in CUCM and dial-peer on Voice gateway.

1) Can you confirm if you have following under your dial-peer:

progress_ind alert enable 8

progress_ind connect enable 8

2) If this doesnot work you can  also try changing the Service parametr in CUCM -  "Send H225 User Info Message" to "Use ANN for rignback". Then assign the  annunciator to a MRG and MRGL - assign the MRGL to both gateway and  phones. See if this makes any difference.

3) Still doesn't works, post your config and also the output of debug voip ccapi inout and debug ccsip messages.

Terry

View solution in original post

+5 aokanlawon spot on.

Tiger - Please follow the above advice and present these logs to your provider.

Normal call flow (simplified) should be - INVITE <> 100 - TRYING <> 180/183 (Ringback happens here) 200- OK -> ACK

You are not receiving ringback from your provider.

Terry

View solution in original post

10 Replies 10

Terry Cheema
VIP Alumni
VIP Alumni

To confirm you are dialing from IP phone to PSTN. What type of Voice gateway are you using for PSTN connectivity SIP/H323/MGCP?

If its an h323 check if you have "progress_ind alert enable 8" command on your voice gateway, under the outbound dial-peer. If its not there try adding and see how it goes.Progress indicator of 8 means inband info is available.

Terry

Hi Terry,

Thanks for the response.

Below the call flow.

Gateway(Router)----H323----CUCM----SIP----Provider(SIP Server)

Regards,

Malik

Can I say that:

From CUCM you are using h323 gateway and from h323 gateway you are using SIP to provider?

CUCM <>H323<>SIP

Terry

Sent from Cisco Technical Support iPhone App

islam.kamal
Level 10
Level 10

Dear

*The terminating switch should send ringback tone if originating gateway signals PI=8. The PI information is then sent thru H225 progress message. The ISDN switch sends back an in−band ringback, but the Alert message does not contain a PI.

check the below link

http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a0080094c33.shtml

Add the following to your config

Please use the below configuation

dial-peer voice 70 pots

preference 1

incoming called-number .

destination-pattern 0T

progress_ind alert enable 8

progress_ind progress enable 8

progress_ind connect enable 8

direct-inward-dial

port 2/0:15

forward-digits all

Thank you ,

please rate if this will help

tiger_401
Level 1
Level 1

Dear Terry,
Yes exactly you are right.

Sent from Cisco Technical Support iPhone App

Hi - First of all, I think its generaly a good idea to get rid of the h323 leg and replace it with SIP. All you would need to do is configure SIP trunk in CUCM and dial-peer on Voice gateway.

1) Can you confirm if you have following under your dial-peer:

progress_ind alert enable 8

progress_ind connect enable 8

2) If this doesnot work you can  also try changing the Service parametr in CUCM -  "Send H225 User Info Message" to "Use ANN for rignback". Then assign the  annunciator to a MRG and MRGL - assign the MRGL to both gateway and  phones. See if this makes any difference.

3) Still doesn't works, post your config and also the output of debug voip ccapi inout and debug ccsip messages.

Terry

Dear Terry,

Thanks a lot for your response. I really appreciate.

I have tested the steps mentioned in your post.

Kindly find attached debug commands for your consideration.

Regards,

Malik.

Tiger,

I have looked at your traces and this is definitely a problemw ith your provider..

+++++++++++CUBE sent first Invite, it dont nor get a trying or a 180 ringing from ITSP++++++++++++++

Sent:
INVITE sip:0566835775@10.200.7.157:5060 SIP/2.0
Via: SIP/2.0/UDP 10.66.4.150:5060;branch=z9hG4bK76AB3
Remote-Party-ID: "ABDULNASSER AL-FANTOUKH" <2835101>;party=calling;screen=yes;privacy=off
From: "ABDULNASSER AL-FANTOUKH" <2835101>;tag=AC78F08-664
To: <0566835775>
Date: Mon, 18 Feb 2013 11:21:58 GMT
Call-ID: 3E50C621-78F411E2-8EC7C531-AB7300C6@10.66.4.150
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 0015293514-3283751186-095632
RYD-MOL-AKA-VGW1#6913-0168509211
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1361186518
Contact: <2835101>
Expires: 180
Allow-Events: telephone-event
Content-Length: 0


006096: Feb 18 11:21:59.237: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

++++++++Cube Sent a second invite and dint get a response again ++++++++++++++
Sent:
INVITE sip:0566835775@10.200.7.157:5060 SIP/2.0
Via: SIP/2.0/UDP 10.66.4.150:5060;branch=z9hG4bK76AB3
Remote-Party-ID: "ABDULNASSER AL-FANTOUKH" <2835101>;party=calling;screen=yes;privacy=off
From: "ABDULNASSER AL-FANTOUKH" <2835101>;tag=AC78F08-664
To: <0566835775>
Date: Mon, 18 Feb 2013 11:21:59 GMT
Call-ID: 3E50C621-78F411E2-8EC7C531-AB7300C6@10.66.4.150
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 0015293514-3283751186-0956326913-0168509211
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1361186519
Contact: <2835101>
Expires: 180
Allow-Events: telephone-event
Content-Length: 0

++++++++CUBE sends a 3rd Invite++++++++++
006097: Feb 18 11:22:00.237: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:0566835775@10.200.7.157:5060 SIP/2.0
Via: SIP/2.0/UDP 10.66.4.150:5060;branch=z9hG4bK76AB3
Remote-Party-ID: "ABDULNASSER AL-FANTOUKH" <2835101>;party=calling;screen=yes;privacy=off
From: "ABDULNASSER AL-FANTOUKH" <2835101>;tag=AC78F08-664
To: <0566835775>
Date: Mon, 18 Feb 2013 11:22:00 GMT
Call-ID: 3E50C621-78F411E2-8EC7C531-AB7300C6@10.66.4.150
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 0015293514-3283751186-0956326913-0168509211
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1361186520
Contact: <2835101>
Expires: 180
Allow-Events: telephone-event
Content-Length: 0


006098: Feb 18 11:22:02.237: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

+++++++++=CUBE sends a 4th Invite, still no response from your ITSP++++++++++++
Sent:
INVITE sip:0566835775@10.200.7.157:5060 SIP/2.0
Via: SIP/2.0/UDP 10.66.4.150:5060;branch=z9hG4bK76AB3
Remote-Party-ID: "ABDULNASSER AL-FANTOUKH" <2835101>;party=calling;screen=yes;privacy=off
From: "ABDULNASSER AL-FANTOUKH" <2835101>;tag=AC78F08-664
To: <0566835775>RYD-MOL-AKA-VGW1#.200.7.157>
Date: Mon, 18 Feb 2013 11:22:02 GMT
Call-ID: 3E50C621-78F411E2-8EC7C531-AB7300C6@10.66.4.150
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 0015293514-3283751186-0956326913-0168509211
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1361186522
Contact: <2835101>
Expires: 180
Allow-Events: telephone-event
Content-Length: 0


006099: Feb 18 11:22:02.921: %IP-4-DUPADDR: Duplicate address 10.66.4.150 on GigabitEthernet0/0, sourced by a44c.1137.6080

++++++++CUBE sends a 5th Invite++++++++++++++++
006100: Feb 18 11:22:06.237: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:0566835775@10.200.7.157:5060 SIP/2.0
Via: SIP/2.0/UDP 10.66.4.150:5060;branch=z9hG4bK76AB3
Remote-Party-ID: "ABDULNASSER AL-FANTOUKH" <2835101>;party=calling;screen=yes;privacy=off
From: "ABDULNASSER AL-FANTOUKH" <2835101>;tag=AC78F08-664
To: <0566835775>
Date: Mon, 18 Feb 2013 11:22:06 GMT
Call-ID: 3E50C621-78F411E2-8EC7C531-AB7300C6@10.66.4.150
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 0015293514-3283
RYD-MOL-AKA-VGW1#751186-0956326913-0168509211
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1361186526
Contact: <2835101>
Expires: 180
Allow-Events: telephone-event
Content-Length: 0

++++++++++++++++All of a sudden we get a 200 ok from your provider +++++++++++++++
RYD-MOL-AKA-VGW1#
006101: Feb 18 11:22:12.857: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.66.4.150:5060;branch=z9hG4bK76AB3
Record-Route: <10.200.7.157:5060>
Call-ID: 3E50C621-78F411E2-8EC7C531-AB7300C6@10.66.4.150
From: "ABDULNASSER AL-FANTOUKH"<2835101>;tag=AC78F08-664
To: <0566835775>;tag=sbc08022t4cpbpc-CC-47
CSeq: 101 INVITE
Contact: <0566835775>
Content-Length: 371
Content-Type: application/sdp

v=0
o=- 16123288 16123290 IN IP4 10.200.7.157
s=SBC call
c=IN IP4 10.200.7.157
t=0 0
m=audio 18394 RTP/AVP 8 0 18 4 2 98 99 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:98 G726-40/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=fmtp:18 annexb=yes

So from the traces, your provider is not sending mandatory sip messages for you to hear ringback. It needs to atleast send a 180 ringing or a 183 session progress

Go back to them and show them these logs. Get them to fix the problem.

I will also echo Terry's advice, use a sip to sip internetworking. Configure CUCM to talk to CUBE using sip and not h323.

Please rate all useful posts

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

Please rate all useful posts

+5 aokanlawon spot on.

Tiger - Please follow the above advice and present these logs to your provider.

Normal call flow (simplified) should be - INVITE <> 100 - TRYING <> 180/183 (Ringback happens here) 200- OK -> ACK

You are not receiving ringback from your provider.

Terry

Thanks to everyone

for your help and support