One way audio related to SIP Invite?

I’ve put the log sequence for a call setup which ends up with one way audio. Here are the details:

10.232.4.4 is an Asterisk PBX
10.232.4.5 is an IAX2 trunk
10.232.4.31 is a Cisco VOIP phone
10.28.10.50 is a SIP trunk to a Cisco Call Manager
"4002" is the number for the phone and is registered to the PBX
"16135561001" is the destination phone registered to the CCM

“4002” sends RTP to the PBX which forwards them over the SIP trunk. “16135561001” send RTP to 10.232.4.5 which “4002” does not hear.

From the logs below, the SIP Invite comes from the right place but then another invite gets sent with a different “from” IP. This IP is my IAX2 trunk.

If I am missing any other details, please let me know.


INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.232.4.31:5060;branch=z9hG4bK50b3f059
From: “4002” sip:[email protected];tag=000b5ff9349505161412ab13-307a0638
To: sip:[email protected]
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 INVITE
User-Agent: Cisco-CP7940G/8.0
Contact: sip:[email protected]:5060;transport=udp
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Supported: replaces,join,norefersub
Content-Length: 274
Content-Type: application/sdp
Content-Disposition: session;handling=optional

3SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.232.4.31:5060;branch=z9hG4bK50b3f059;received=10.232.4.31
From: “4002” sip:[email protected];tag=000b5ff9349505161412ab13-307a0638
To: sip:[email protected];tag=as3093ade2
Call-ID: [email protected]
CSeq: 101 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="0d31ed68"
Content-Length: 0

ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.232.4.31:5060;branch=z9hG4bK50b3f059
From: “4002” sip:[email protected];tag=000b5ff9349505161412ab13-307a0638
To: sip:[email protected];tag=as3093ade2
Call-ID: [email protected]
CSeq: 101 ACK
Content-Length: 0

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.232.4.31:5060;branch=z9hG4bK0e3bc5f6
From: “4002” sip:[email protected];tag=000b5ff9349505161412ab13-307a0638
To: sip:[email protected]
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 INVITE
User-Agent: Cisco-CP7940G/8.0
Contact: sip:[email protected]:5060;transport=udp
Authorization: Digest username=“4002”,realm=“asterisk”,uri="sip:[email protected]",response=“84fca2ac3180a23c5b90d3eb2b646981”,nonce=“0d31ed68”,algorithm=MD5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Supported: replaces,join,norefersub
Content-Length: 274
Content-Type: application/sdp
Content-Disposition: session;handling=optional

Trying
Via: SIP/2.0/UDP 10.232.4.31:5060;branch=z9hG4bK0e3bc5f6;received=10.232.4.31
From: “4002” sip:[email protected];tag=000b5ff9349505161412ab13-307a0638
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Length: 0

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.232.4.5:5060;branch=z9hG4bK148dea05
Max-Forwards: 70
From: sip:[email protected];tag=as0d3576fd
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.13.0)
Date: Thu, 28 Jul 2016 13:43:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 232

Trying
Via: SIP/2.0/UDP 10.232.4.5:5060;branch=z9hG4bK148dea05;received=10.232.4.4
From: sip:[email protected];tag=as0d3576fd
To: sip:[email protected]
Date: Thu, 28 Jul 2016 14:32:30 GMT
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.4.2.T
Content-Length: 0

Ringing
Via: SIP/2.0/UDP 10.232.4.5:5060;branch=z9hG4bK148dea05;received=10.232.4.4
From: sip:[email protected];tag=as0d3576fd
To: sip:[email protected];tag=BFFD5248-235D
Date: Thu, 28 Jul 2016 14:32:30 GMT
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: “Test01” sip:[email protected];party=called;screen=no;privacy=off
Contact: sip:[email protected]:5060
Server: Cisco-SIPGateway/IOS-15.4.2.T
Content-Length: 0

Ooops, I see the “To:” field was strangely omitted from my log pastes. Actually this is interesting and might be important. Here is the field for the first invites:

To:sip:[email protected]
Call-ID: [email protected]

And here it is for the second invite which is different and now has a “CONTACT” line:

To: sip:[email protected]
Contact: sip:[email protected]:5060

Note that “CALL ID” has changed.

I’m not sure why my pasting is not working properly.

Make a test call and post here what you get in the cli.

That’s what the log I posted is, an example call.