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New Member

CUBE SIP Trunk with Callcentric - Inbound call not answered

I am doing some testing with a Sip trunk with a Provider called Callcentric.
It is a CUBE scenario. I use a SIP trunk to the CUCM.

I have a Cisco Callmanager and a virtual router 7200 (GNS3) as a gateway (C7200-ADVENTERPRISEK9-M), Version 12.4(24)

From a CIPC connected to Callmanager I make an outbound call to the PSTN and it works perfectly. 

When I make an inbound call from the PSTN to the softphone, it rings but when I press Answer button, the call is not connected and the analog phone in the PSTN keeps listening ring back tone. 

Do you have any idea what it could be?? 

 
Some relevant configurations:
 

voice service voip
 allow-connections sip to sip
 fax protocol cisco
 sip
  bind control source-interface FastEthernet0/0
  bind media source-interface FastEthernet0/0
  registrar server

voice class codec 1
 codec preference 1 g711ulaw

voice translation-rule 1
 rule 1 /^8/ /0056/
!
voice translation-rule 2
 rule 1 /5../ /17772114zzz/
!
voice translation-rule 3
 rule 1 /17772114zzz/ /500/

 

voice translation-profile IN
 translate called 3
!
voice translation-profile OUT
 translate calling 2
 translate called 1

 

 

dial-peer voice 1 voip
 description CALLCENTRIC
 translation-profile incoming IN
 translation-profile outgoing OUT
 destination-pattern 8.T
 voice-class codec 1
 session protocol sipv2
 session target sip-server
 incoming called-number 17772114zzz
 dtmf-relay sip-notify rtp-nte
!
dial-peer voice 2 voip
 description CUCM
 destination-pattern 500
 media flow-around
 voice-class codec 1
 session protocol sipv2
 session target ipv4:192.168.10.116
 incoming called-number 8.T
 dtmf-relay sip-notify rtp-nte

 

sip-ua
 credentials username 17772114zzz password 7 094F471B100928 realm callcentric.com
 authentication username 17772114zzz password 7 121A0C051B0803 realm callcentric.com
 no remote-party-id
 registrar dns:callcentric.com expires 3600
 sip-server dns:callcentric.com
 host-registrar

 


Thanks guys.
 
 
Everyone's tags (1)
1 ACCEPTED SOLUTION

Accepted Solutions

HiPlease configure the below

Hi

Please configure the below sip profile to remove the Require Header and Apply this profile in the outgoing dial-peer 1.

voice class sip-profiles 1

response 200 sip-header Require REMOVE

 

if that doesn't work under dial-peers, you may try applying it globally.

voice service voip

sip

sip-profiles 1

 

//Suresh

Please rate all the helpful posts

//Suresh Please rate all the useful posts.
9 REPLIES

Hi,you may need to create

Hi,

you may need to create separate Inbound & outbound dial-peers for both incoming & outgoing direction, total of 4 dial-peers.

 

if the issue persists, could you please collect debug ccsip message & debug voice ccapi inout for a call?

 

//Suresh

Please rate all the helpful posts

//Suresh Please rate all the useful posts.
New Member

Well, actually in the first

Well, actually in the first time I ´ve got 4 dial-peers.

1 inbound from SIP Provider

1 outbound to SiP Provider

1 inbound from CUCM

1 outbound to CUCM

and the problem was the same.

Afterwards I simplify the configuration in 2 dial-peers like I showed in previous post but the same result.

I attached the debugs for an inbound call. Extension 500 (CIPC) rings, I press Answer button but the analog phone at the PSTN keeps listening ring back tone. Finally CIPS ends  the call.

I tested this using CME in the router itself. It works perfect for inbound and outbound calls. Obviously in this case dial-peer to CUCM doesn´t exist.

 

Any ideas to solve this??

 

Thank you.

Hi, is this the issue only

Hi, is this the issue only with CIPC or Hardphones also? From the debug, it seems there is a call forward to 8971626277. Please check that. Also in the CIPC settings, "optimize for low bandwidth" is checked? If yes, please uncheck it and try.
//Suresh Please rate all the useful posts.
New Member

I also tested with a 3CX SIP

I also tested with a 3CX SIP softphone and the problem was the same. The call forward you see is actually because of  a Single Number Reach Configuration, but the problem remains without that functionality. All call legs are using G711ulaw (optimize for low bandwidth disabled).

I have not tried with a physical phone but I use CIPC in a Call Manager Express environment with the same SIP Provider and works great.

I think it is a problem of signaling. When I end the call from the CIPC, el analog phone in the PSTN keeps with ring back tone. When I end the call from the analog phone, CIPS ends the call also. So the signaling to the PSTN is like not passing for an inbound call, but the signaling from the PSTN can pass ok for an inbound call.

Thank you.

 

 

Hi,

Hi,

I checked the attached debugs and seems the issue is with provider side who is not responding to the 200 OK message sent from CUBE.

>> Invite message sent to CUCM

*May 24 15:35:04.515: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:500@192.168.10.116:5060 SIP/2.0
Date: Sat, 24 May 2014 15:35:04 GMT
Call-Info: <sip:192.168.10.10:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
From: <sip:56222263788@callcentric.com>;tag=C18E38-10E6
Allow-Events: telephone-event
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 3483623425-3801027043-2167054572-2285343397
Timestamp: 1400945704
Content-Length: 274
User-Agent: Cisco-SIPGateway/IOS-12.x
To: <sip:500@192.168.10.116>
Contact: <sip:56222263788@192.168.10.10:5060>
Expires: 180
Content-Disposition: session;handling=required
Content-Type: application/sdp
Call-ID: CFB9D89B-E28F11E3-8130A0EC-883792A5@192.168.10.10
Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK15A52E
CSeq: 101 INVITE
Max-Forwards: 12

v=0
o=CiscoSystemsSIP-GW-UserAgent 6959 7987 IN IP4 192.168.10.10
s=SIP Call
c=IN IP4 204.11.192.39
t=0 0
m=audio 60992 RTP/AVP 0 101 19
c=IN IP4 204.11.192.39
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
a=ptime:20


>> 100 Trying received from CUCM

*May 24 15:35:04.563: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK15A52E
From: <sip:56222263788@callcentric.com>;tag=C18E38-10E6
To: <sip:500@192.168.10.116>
Date: Sat, 24 May 2014
CUBE#19:35:19 GMT
Call-ID: CFB9D89B-E28F11E3-8130A0EC-883792A5@192.168.10.10
CSeq: 101 INVITE
Allow-Events: presence
Content-Length: 0

>> 180 Ringing Received from CUCM

*May 24 15:35:04.579: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK15A52E
From: <sip:56222263788@callcentric.com>;tag=C18E38-10E6
To: <sip:500@192.168.10.116>;tag=286~ff02ad80-68cc-4826-be4d-e6d9992b00f8-16971568
Date: Sat, 24 May 2014 19:35:19 GMT
Call-ID: CFB9D89B-E28F11E3-8130A0EC-883792A5@192.168.10.10
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Allow-Events: presence
Call-Info: <sip:192.168.10.116:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Supported: X-cisco-srtp-fallback
Supported: Geolocation
P-Asserted-Identity: "Mario Garay CIPT" <sip:500@192.168.10.116>
Remote-Party-ID: "Mario Garay CIPT" <sip:500@192.168.10.116>;party=called;screen=yes;privacy=off
Contact: <sip:500@192.168.10.116:5060>
Content-Length: 0

>> call is answered and received 200 OK for that.

*May 24 15:35:08.559: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK15A52E
From: <sip:56222263788@callcentric.com>;tag=C18E38-10E6
To: <sip:500@192.168.10.116>;tag=286~ff02ad80-68cc-4826-be4d-e6d9992b00f8-16971568
Date: Sat, 24 May 2014 19:35:19 GMT
Call-ID: CFB9D89B-E28F11E3-8130A0EC-883792A5@192.168.10.10
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Allow-Events: presence, kpml
Supported: replaces
Call-Info: <sip:192.168.10.116:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires:  1800;refresher=uas
Require:  timer
P-Asserted-Identity: "Mario Garay" <sip:500@192.168.10.116>
Remote-Party-ID: "Mario Garay" <sip:500@192.168.10.116>;party=called;screen=yes;privacy=off
Contact: <sip:500@192.168.10.116:5060>
Content-Type: application/sdp
Content-Length: 160

v=0
o=CiscoSystemsCCM-SIP 2000 1 IN IP4 192.168.10.116
s=SIP Call
c=IN IP4 192.168.10.116
t=0 0
m=audio 24742 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20

>> CUBE acknowledged the same

*May 24 15:35:08.667: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:500@192.168.10.116:5060 SIP/2.0
Date: Sat, 24 May 2014 15:35:04 GMT
From: <sip:56222263788@callcentric.com>;tag=C18E38-10E6
Allow-Events: telephone-event
Content-Length: 0
To: <sip:500@192.168.10.116>;tag=286~ff02ad80-68cc-4826-be4d-e6d9992b00f8-16971568
Call-ID: CFB9D89B-E28F11E3-8130A0EC-883792A5@192.168.10.10
Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK15DD01
CSeq: 101 ACK
Max-Forwards: 70

>> when the CIPC answers the calls, CUCM sent 200 OK message to CUBE and it acknowledged it by sending ACK back to CUCM.

>> now CUBE forwarded the 200 OK message to ITSP but no response from ITSP side.

>> CUBE retried sending the 200 OK message for 4 more times but no response.

>> 1st 200 OK sent to ITSP

*May 24 15:35:08.787: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Date: Sat, 24 May 2014 15:35:04 GMT
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
From: <sip:56222263788@66.193.176.35:5060>;tag=as516768f4
Allow-Events: telephone-event
Supported: replaces
Supported: sdp-anat
Content-Length: 250
Require: timer
To: <sip:56225814887@ss.callcentric.com>;tag=C18EC8-55A
Contact: <sip:17772114441@192.168.10.10:5060>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Call-ID: 32559519-3609948918-834417@msw2.telengy.net
Via: SIP/2.0/UDP 204.11.192.39:5080;branch=z9hG4bK-5ed830c214e955b748004295dd4666b7;change=ta
CSeq: 1 INVITE
Server: Cisco-SIPGateway/IOS-12.x
Session-Expires:  1800;refresher=uas

v=0
o=CiscoSystemsSIP-GW-UserAgent 8260 1252 IN IP4 192.168.10.10
s=SIP Call
c=IN IP4 192.168.10.10
t=0 0
m=audio 29258 RTP/AVP 0 101
c=IN IP4 192.168.10.10
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

>> 2nd 200 OK sent to ITSP

*May 24 15:35:09.307: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Date: Sat, 24 May 2014 15:35:04 GMT
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
From: <sip:56222263788@66.193.176.35:5060>;tag=as516768f4
Allow-Events: telephone-event
Supported: replaces
Supported: sdp-anat
Content-Length: 250
Require: timer
To: <sip:56225814887@ss.callcentric.com>;tag=C18EC8-55A
Contact: <sip:17772114441@192.168.10.10:5060>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Call-ID: 32559519-3609948918-834417@msw2.telengy.net
Via: SIP/2.0/UDP 204.11.192.39:5080;branch=z9hG4bK-5ed830c214e955b748004295dd4666b7;change=ta
CSeq: 1 INVITE
Server: Cisco-SIPGateway/IOS-12.x
Session-Expires:  1800;refresher=uas

v=0
o=CiscoSystemsSIP-GW-UserAgent 8260 1252 IN IP4 192.168.10.10
s=SIP Call
c=IN IP4 192.168.10.10
t=0 0
m=audio 29258 RTP/AVP 0 101
c=IN IP4 192.168.10.10
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

>> 3rd 200 OK sent to ITSP

*May 24 15:35:10.315: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Date: Sat, 24 May 2014 15:35:04 GMT
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
From: <sip:56222263788@66.193.176.35:5060>;tag=as516768f4
Allow-Events: telephone-event
Supported: replaces
Supported: sdp-anat
Content-Length: 250
Require: timer
To: <sip:56225814887@ss.callcentric.com>;tag=C18EC8-55A
Contact: <sip:17772114441@192.168.10.10:5060>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Call-ID: 32559519-3609948918-834417@msw2.telengy.net
Via: SIP/2.0/UDP 204.11.192.39:5080;branch=z9hG4bK-5ed830c214e955b748004295dd4666b7;change=ta
CSeq: 1 INVITE
Server: Cisco-SIPGateway/IOS-12.x
Session-Expires:  1800;refresher=uas

v=0
o=CiscoSystemsSIP-GW-UserAgent 8260 1252 IN IP4 192.168.10.10
s=SIP Call
c=IN IP4 192.168.10.10
t=0 0
m=audio 29258 RTP/AVP 0 101
c=IN IP4 192.168.10.10
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20


>> 4th 200 OK sent to ITSP

*May 24 15:35:12.319: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Date: Sat, 24 May 2014 15:35:04 GMT
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
From: <sip:56222263788@66.193.176.35:5060>;tag=as516768f4
Allow-Events: telephone-event
Supported: replaces
Supported: sdp-anat
Content-Length: 250
Require: timer
To: <sip:56225814887@ss.callcentric.com>;tag=C18EC8-55A
Contact: <sip:17772114441@192.168.10.10:5060>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Call-ID: 32559519-3609948918-834417@msw2.telengy.net
Via: SIP/2.0/UDP 204.11.192.39:5080;branch=z9hG4bK-5ed830c214e955b748004295dd4666b7;change=ta
CSeq: 1 INVITE
Server: Cisco-SIPGateway/IOS-12.x
Session-Expires:  1800;refresher=uas

v=0
o=CiscoSystemsSIP-GW-UserAgent 8260 1252 IN IP4 192.168.10.10
s=SIP Call
c=IN IP4 192.168.10.10
t=0 0
m=audio 29258 RTP/AVP 0 101
c=IN IP4 192.168.10.10
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

>> 5th 200 OK sent to ITSP

*May 24 15:35:16.315: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Date: Sat, 24 May 2014 15:35:04 GMT
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
From: <sip:56222263788@66.193.176.35:5060>;tag=as516768f4
Allow-Events: telephone-event
Supported: replaces
Supported: sdp-anat
Content-Length: 250
Require: timer
To: <sip:56225814887@ss.callcentric.com>;tag=C18EC8-55A
Contact: <sip:17772114441@192.168.10.10:5060>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Call-ID: 32559519-3609948918-834417@msw2.telengy.net
Via: SIP/2.0/UDP 204.11.192.39:5080;branch=z9hG4bK-5ed830c214e955b748004295dd4666b7;change=ta
CSeq: 1 INVITE
Server: Cisco-SIPGateway/IOS-12.x
Session-Expires:  1800;refresher=uas

v=0
o=CiscoSystemsSIP-GW-UserAgent 8260 1252 IN IP4 192.168.10.10
s=SIP Call
c=IN IP4 192.168.10.10
t=0 0
m=audio 29258 RTP/AVP 0 101
c=IN IP4 192.168.10.10
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20


>> BYE received from CUCM side
*May 24 15:35:12.787: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:56222263788@192.168.10.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.116:5060;branch=z9hG4bK98239541cb
From: <sip:500@192.168.10.116>;tag=286~ff02ad80-68cc-4826-be4d-e6d9992b00f8-16971568
To: <sip:56222263788@callcentric.com>;tag=C18E38-10E6
Date: Sat, 24 May 2014 19:35:23 GMT
Call-ID: CFB9D89B-E28F11E3-8130A0EC-883792A5@192.168.10.10
User-Agent: Cisco-CUCM8.5
Max-Forwards: 70
CSeq: 101 BYE
Content-Length: 0

>> CUBE acknowledge the BYE message
*May 24 15:35:12.863: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Reason: Q.850;cause=16
Date: Sat, 24 May 2014 15:35:12 GMT
From: <sip:500@192.168.10.116>;tag=286~ff02ad80-68cc-4826-be4d-e6d9992b00f8-16971568
Content-Length: 0
To: <sip:56222263788@callcentric.com>;tag=C18E38-10E6
Call-ID: CFB9D89B-E28F11E3-8130A0EC-883792A5@192.168.10.10
Via: SIP/2.0/UDP 192.168.10.116:5060;branch=z9hG4bK98239541cb
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 BYE

>> ITSP sent BYE message to CUBE

*May 24 15:35:16.723: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:17772114441@192.168.10.10:5060 SIP/2.0
v: SIP/2.0/UDP 204.11.192.39:5080;branch=z9hG4bK-87824e3b093ab808a215b8725b00d465;change=ta
f: <sip:56222263788@66.193.176.35:5060>;tag=as516768f4
t: <sip:56225814887@ss.callcentric.com>;tag=C18EC8-55A
i: 32559519-3609948918-834417@msw2.telengy.net
CSeq: 2 BYE
Max-Forwards: 13
l: 0


>> I request you to talk to your provider on this issue.

 

//Suresh

Please rate all the helpful posts

//Suresh Please rate all the useful posts.
New Member

OK, I am reviewing your

OK, I am reviewing your analysis. I will notice Callcentric about the problem.

Thank you so much for the analysis.

 

 

New Member

Hello, that is the ITSP

Hello, that is the ITSP Callcentric responded about:

--------------------------------------------------------------------------------

SIP/2.0 200 OK
Date: Sun, 25 May 2014 19:24:54 GMT
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
From: <sip:5101100061612@66.193.176.35:5060>;tag=as34df9acb
Allow-Events: telephone-event
Supported: replaces
Supported: sdp-anat
Content-Length: 250
Require: timer
To: <sip:56225814887@ss.callcentric.com>;tag=240030-716
Contact: <sip:17772114441@192.168.10.10:5060>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Call-ID: 32826171-3610049108-400787@msw2.telengy.net
Via: SIP/2.0/UDP 204.11.192.22:5080;branch=z9hG4bK-d68ba020764b2d5f23e1ab8142c1636b;change=ta
CSeq: 1 INVITE
Server: Cisco-SIPGateway/IOS-12.x
Session-Expires: 1800;refresher=uas

v=0
o=CiscoSystemsSIP-GW-UserAgent 9525 7588 IN IP4 192.168.10.10
s=SIP Call
c=IN IP4 192.168.10.10
t=0 0
m=audio 23304 RTP/AVP 0 101
c=IN IP4 192.168.10.10
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

 

Basically what's occurring above is that while our test call attempt is sent towards your Cisco SIP gateway, and your SIP gateway is receiving it; it isn't established properly, specifically as your gateway seems to be sending the header "Require: timer" which isn't necessary (we believe this is the reason why calls are not being established properly). If possible, can you configure your gateway to remove this particular header?

-------------------------------------------------------------------------

Is there any command to remove that option at the gateway?

 

Thanks.

 

HiPlease configure the below

Hi

Please configure the below sip profile to remove the Require Header and Apply this profile in the outgoing dial-peer 1.

voice class sip-profiles 1

response 200 sip-header Require REMOVE

 

if that doesn't work under dial-peers, you may try applying it globally.

voice service voip

sip

sip-profiles 1

 

//Suresh

Please rate all the helpful posts

//Suresh Please rate all the useful posts.
New Member

Solved. It works with the SIP

Solved. It works with the SIP Profile.

Many Thanks.

For CallCentric SIP Trunk, a SIP Profile is a must.

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