SIP TRUNK ITSP Error Code 65 - Bearer Capability Not Implemented

[Franklin Gonzalez Sevillano]

SIP TRUNK ITSP Error Code 65 - Bearer Capability Not Implemented with the following call flow.

PHONE ----CUCM 9.1.2 ----------SIP TRUNK (DO)---------- ISR 2801 151-4.M12a ----------SIP TRUNK (EO)---------- ITSP

Media Termination Point Required over SIP TRUNK

Codec g711alaw Throughout the flow of the call

Outgoing calls are set well, however when you put the call on hold and then try to resume the call. The call drops, this only happens with outgoing calls. Attachment file (debug ccsip all), I have the same scenario in other clients and the same ITSP but with CUBE (ISR2) and it works without problems.

Thanks!

[Mohammed al Baqari]

Not sure about your config but paste this command to disable call refer.

voice service voip

no supplementary-service sip moved-temp

[Franklin Gonzalez Sevillano]

Hi Mohammed,

Thanks for your answer. I already had the command applied:

My voice service voip config:

voice service voip
 address-hiding
 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
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 sip
  header-passing
  error-passthru
  registrar server expires max 3600 min 600
  early-offer forced
  midcall-signaling passthru

[Mohammed al Baqari]

Hi,

In this case, please share your full debug. You messages aren't showing any INVITE received from CUCM to hold the call. All messages showed are ones generated by CUBE. 

Also, please try to enable antitromboning under your voice service voip and see if it fixes the problem. 

[Franklin Gonzalez Sevillano]

The media anti-trombone command is enabled by default. Certainly the debug does not show the complete interaction between CUBE and CUCM, although I have taken it again with the same procedure without success.

Annex procedure:

service sequence-numbers
service timestamps debug datetime localtime msec
logging buffered 10000000 debug
no logging console
no logging monitor
default logging rate-limit
default logging queue-limit
 
Then..
 
<Enable debugs, then test again.>
 
debug ccsip all
 
<Enable session capture to txt file in terminal program.>
 
 
then do the ff:
 
terminal length 0
show logging

[HARIS_HUSSAIN]

From the logs i can see that the Port Negotiation failed as the Provided send 0 as the port address for audio.

Received: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.1.1:5060;branch=z9hG4bK43321D25
From: <sip:223622708@172.16.1.1;user=phone>;tag=404C6D90-153A
To: <sip:6005100000@172.22.25.51;user=phone;user=phone;user=phone;user=phone;user=phone>;tag=tspqrp3l-CC-1004
Call-ID: DB2EACB0-17BD11E7-88619271-618046DE@172.16.1.1
CSeq: 107 INVITE
Timestamp: 1491235399
Contact: <sip:6005100000@172.22.25.51:5060;transport=udp>
Content-Length: 203
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 237274 237278 IN IP4 172.22.25.51
s=Sip Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 0 RTP/AVP 8 97
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=inactive

001628: Apr 3 13:03:20.057: //20926/009EB95C1701/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 172.16.1.1
001629: Apr 3 13:03:20.057: //20926/009EB95C1701/SIP/Info/sipSPIValidateConnectionAddress: dead stream since destination port is 0 for the m-line : 1
001630: Apr 3 13:03:20.057: //20926/009EB95C1701/SIP/Error/sipSPIDoMediaNegotiation: 
no valid fax or audio streams
001631: Apr 3 13:03:20.057: //20926/009EB95C1701/SIP/Error/ccsip_api_response_answer: Media Negotiation failure in 200 OK

[Franklin Gonzalez Sevillano]

What do you think this situation should be? In the same CUBE we have connected another ITSP with which this problem does not occur1