Can't make outbound calls to sip provider. Inbound calling works fine

[jcajuste]

I am having an issue with outbound sip calls.  Inbound calls works fine but anytime we try to make an outbound call the phone goes to fast busy.  attached are  debug ccsip message
debug voip ccapi inout

[Sadav Ansari · Accepted solution]

Hi,

As per the debug

---

Sent Invite msg 

000990: May 22 16:10:25.406 EDT: //63267/8F6079B897B9/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:13013000894@us-west-or.sip.flowroute.com:5060 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK828E15E6
From: "Conference RoomOne" <sip:13012325675@us-west-or.sip.flowroute.com>;tag=5894DB20-E03
To: <sip:13013000894@us-west-or.sip.flowroute.com>
Date: Sat, 22 May 2021 20:10:25 GMT
Call-ID: 945B8A52-BA7011EB-97BF98FC-521C789A@us-west-or.sip.flowroute.com
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 2405464504-3127906795-2545522940-1377597594
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1621714225
Contact: <sip:13012325675@X.X.X.X:5060>
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="XXXXXX",realm="sip.flowroute.com",uri="sip:13013000894@us-west-or.sip.flowroute.com:5060",response="f171d378514a4c65687c3a7bd6760b39",nonce="YKltgWCpbFXBzTTQZwaVwwpBt5ulns1W",cnonce="FE7430F4",qop=auth,algorithm=md5,nc=00000001
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 273

v=0
o=CiscoSystemsSIP-GW-UserAgent 8126 1618 IN IP4 XXX.XXX.XXX.XXX
s=SIP Call
c=IN IP4 X.X.X.X
t=0 0
m=audio 17346 RTP/AVP 18 101
c=IN IP4 XXX.XXX.XXX.XXX
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

Gateway sent 100 trying to          originator

000991: May 22 16:10:25.494 EDT: //63267/8F6079B897B9/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK828E15E6;received=XXX.XXX.XXX.XXX
From: "Conference RoomOne" <sip:13012325675@us-west-or.sip.flowroute.com>;tag=5894DB20-E03
To: <sip:13013000894@us-west-or.sip.flowroute.com>
Call-ID: 945B8A52-BA7011EB-97BF98FC-521C789A@us-west-or.sip.flowroute.com
CSeq: 102 INVITE
Content-Length: 0


000992: May 22 16:10:25.494 EDT: //63267/8F6079B897B9/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport=5060;received=XXX.XXX.XXX.XXX;branch=z9hG4bK828E15E6
From: "Conference RoomOne" <sip:13012325675@us-west-or.sip.flowroute.com>;tag=5894DB20-E03
To: <sip:13013000894@us-west-or.sip.flowroute.com>;tag=bf8638324618dc61059d4c604476fea1.6ca88ad4
Call-ID: 945B8A52-BA7011EB-97BF98FC-521C789A@us-west-or.sip.flowroute.com
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="sip.flowroute.com", nonce="YKltgWCpbFXBzTTQZwaVwwpBt5ulns1W", qop="auth"
Content-Length: 0

Suddenly received call disconnected msg because of cause code 47


000993: May 22 16:10:25.494 EDT: //63267/8F6079B897B9/CCAPI/cc_api_call_disconnected:
   Cause Value=47, Interface=0x877F8F40, Call Id=63267
000994: May 22 16:10:25.494 EDT

Call disconnection cause code is 47.

 In this case cause code 47 is usually a media failure. This is commonly caused by codec mismatch on the VOIP leg.

Cause 47 Resource unavailable, unspecified - This cause is used to report a resource unavailable event only when no other cause in the resource unavailable class applies.

Pls rate if its “Helpful”. If this answered your question pls click “Accept as Solution”.

Sadav Ansari

[Sadav Ansari · Accepted solution]

Hi,

As per the debug.

++Sent Invite++

Sent:

INVITE sip:13013000894@us-west-or.sip.flowroute.com:5060 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK828E15E6
From: "Conference RoomOne" <sip:13012325675@us-west-or.sip.flowroute.com>;tag=5894DB20-E03
To: <sip:13013000894@us-west-or.sip.flowroute.com>
Date: Sat, 22 May 2021 20:10:25 GMT
Call-ID: 945B8A52-BA7011EB-97BF98FC-521C789A@us-west-or.sip.flowroute.com
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2405464504-3127906795-2545522940-1377597594
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1621714225
Contact: <sip:13012325675@X.X.X.X:5060>
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="XXXXXX",realm="sip.flowroute.com",uri="sip:13013000894@us-west-or.sip.flowroute.com:5060",response="f171d378514a4c65687c3a7bd6760b39",nonce="YKltgWCpbFXBzTTQZwaVwwpBt5ulns1W",cnonce="FE7430F4",qop=auth,algorithm=md5,nc=00000001
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 273

v=0
o=CiscoSystemsSIP-GW-UserAgent 8126 1618 IN IP4 XXX.XXX.XXX.XXX
s=SIP Call
c=IN IP4 X.X.X.X
t=0 0
m=audio 17346 RTP/AVP 18 101
c=IN IP4 XXX.XXX.XXX.XXX
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

++ Received 100 Trying from CM ++

000991: May 22 16:10:25.494 EDT: //63267/8F6079B897B9/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK828E15E6;received=XXX.XXX.XXX.XXX
From: "Conference RoomOne" <sip:13012325675@us-west-or.sip.flowroute.com>;tag=5894DB20-E03
To: <sip:13013000894@us-west-or.sip.flowroute.com>
Call-ID: 945B8A52-BA7011EB-97BF98FC-521C789A@us-west-or.sip.flowroute.com
CSeq: 102 INVITE
Content-Length: 0

000992: May 22 16:10:25.494 EDT: //63267/8F6079B897B9/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport=5060;received=XXX.XXX.XXX.XXX;branch=z9hG4bK828E15E6
From: "Conference RoomOne" <sip:13012325675@us-west-or.sip.flowroute.com>;tag=5894DB20-E03
To: <sip:13013000894@us-west-or.sip.flowroute.com>;tag=bf8638324618dc61059d4c604476fea1.6ca88ad4
Call-ID: 945B8A52-BA7011EB-97BF98FC-521C789A@us-west-or.sip.flowroute.com
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="sip.flowroute.com", nonce="YKltgWCpbFXBzTTQZwaVwwpBt5ulns1W", qop="auth"
Content-Length: 0

++Suddenly Received Disconnection mesaage with Cuase Value 47 ++

000993: May 22 16:10:25.494 EDT: //63267/8F6079B897B9/CCAPI/cc_api_call_disconnected:
Cause Value=47, Interface=0x877F8F40, Call Id=63267
000994: May 22 16:10:25.494 EDT: //63267/8F6079B897B9/CCAPI/cc_api_call_disconnected:
Call Entry(Responsed=TRUE, Cause Value=47, Retry Count=0)
000995: May 22 16:10:25.498 EDT: //63265/8F6079B897B9/CCAPI/ccCallReleaseResources:
release reserved xcoding resource.

Cause No. 47 - resource unavailable, unspecified.
This cause is used to report a resource unavailable event only when no other cause in the resource unavailable class applies.

This is generally occured becuse of coded mismacth on Voice call leg so pls check code on CUCM region and on Voice gateway Voice class service.

Pls rate if its " Helpful". If this answered your question pls click " Accept As Solution".

[Sadav Ansari · Accepted solution]

Hi,

As per debug

//63267/8F6079B897B9/CCAPI/ccCallDisconnect:
   Cause Value=47, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=47)
000998: May 22 16:10:25.498 EDT: //63267/8F6079B897B9/CCAPI/ccCallDisconnect:
   Cause Value=47, Call Entry(Responsed=TRUE, Cause Value=47)
000999: May 22 16:10:25.498 EDT: //63267/8F6079B897B9/CCAPI/cc_api_call_disconnect_done:

 
Call disconnection cause value 47.

Cause 47 Resource unavailable, unspecified - This cause is used to report a resource unavailable event only when no other cause in the resource unavailable class applies.

In this case cause code 47 is usually a media failure. This is commonly caused by codec mismatch on the VOIP leg.

Pls review if its “Helpful”. If this answered your question pls click “Accept as Solution”.

[Sadav Ansari]

++Check MTP is checked or not your SIP TRUNK.

Pls rate if its “Helpful”. If this answered your question pls click “Accept as Solution”. 

[jcajuste]

thanks for your help the service provider and the UC could not negotiate g729r8.  change the codec to g711u and calls are working thanks for the help.

[Nithin Eluvathingal]

SIP/2.0 407 Proxy Authentication Required, which interface you bind for media and control ? is it correct interface you used ? Can try the below command under the outgoing dial-peer.

 voice-class sip bind control source-interface <<your Interface connected to ISP>>
 voice-class sip bind media source-interface <<your Interface connected to ISP>>

From the host name i assume this is UC 540. can you share the voice-class codec 1 configuration.

You are sending invite with g729 codec, most ISP prefer g711. Can you also change the codec g711.

[TONY SMITH · Accepted solution]

For codecs you can look at an incoming call, see what codecs the ISP offers in his initial Invite.

On the outgoing call issue, the ITSP is looking for authentication, what do you have configured?  Can you share you full configuration for the gateway?  It looks like you may have configuration for registration but not for authentication.  For example a "credentials username .." line but not "authentication username ..."