New PBX-No Audio and call disconnects after 25 seconds

Hi Team,

We have freshly installed the incredible PBX, however, on calling b/w exts, I am getting error " Retransmission timeout reached on transmission uG3yl-xfLfsJ5aSGwXajJQ… for seqno 2 (Critical Response) --See Packet timed out after 17793ms with no response ".

I also checked the logs after turning Sip debug on but I am not able understand why the re-transmission is happening. I really need to fix this issue/

This is the wrong forum for Incredible PBX, you want this place. That said you have a NAT/network issue and that is why you’re getting timeouts.

+1 on NAT issues due to misconfiguration

Below is the sip debug logs for your reference.

<— SIP read from UDP:10.200.33.126:5066 —>
INVITE sip:[email protected]:5066;transport=UDP SIP/2.0
Via: SIP/2.0/UDP ISP1_Public_IP:5066;branch=z9hG4bK-d8754z-858f22890add152c-1—d8754z-
Max-Forwards: 70
Contact: <sip:[email protected]_Public_IP:5066;transport=UDP>
To: <sip:[email protected]:5066;transport=UDP>
From: “3353”<sip:[email protected]:5066;transport=UDP>;tag=35468a11
Call-ID: M2Q5MDYwMGZjNjYzOThhMzNlMTMxZGI3ZTY0N2RlMWE.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Zoiper for Windows 2.43 r24984
Allow-Events: presence, kpml
Content-Length: 329

v=0
o=Zoiper_user 0 0 IN IP4 ISP1_Public_IP
s=Zoiper_session
c=IN IP4 ISP1_Public_IP
t=0 0
m=audio 8000 RTP/AVP 3 0 8 110 98 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (14 headers 15 lines) —
Sending to 10.200.33.126:5066 (NAT)
Sending to 10.200.33.126:5066 (NAT)
Using INVITE request as basis request - M2Q5MDYwMGZjNjYzOThhMzNlMTMxZGI3ZTY0N2RlMWE.
Found peer ‘3353’ for ‘3353’ from 10.200.33.126:5066

<— Reliably Transmitting (no NAT) to 10.200.33.126:5066 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP ISP1_Public_IP:5066;branch=z9hG4bK-d8754z-858f22890add152c-1—d8754z-;received=10.200.33.126
From: “3353”<sip:[email protected]:5066;transport=UDP>;tag=35468a11
To: <sip:[email protected]:5066;transport=UDP>;tag=as46c5438d
Call-ID: M2Q5MDYwMGZjNjYzOThhMzNlMTMxZGI3ZTY0N2RlMWE.
CSeq: 1 INVITE
Server: FPBX-13.0.195.28(13.28.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“33a37f46”
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘M2Q5MDYwMGZjNjYzOThhMzNlMTMxZGI3ZTY0N2RlMWE.’ in 16448 ms (Method: INVITE)

<— SIP read from UDP:10.200.33.126:5066 —>
ACK sip:[email protected]:5066;transport=UDP SIP/2.0
Via: SIP/2.0/UDP ISP1_Public_IP:5066;branch=z9hG4bK-d8754z-858f22890add152c-1—d8754z-
Max-Forwards: 70
To: <sip:[email protected]:5066;transport=UDP>;tag=as46c5438d
From: “3353”<sip:[email protected]:5066;transport=UDP>;tag=35468a11
Call-ID: M2Q5MDYwMGZjNjYzOThhMzNlMTMxZGI3ZTY0N2RlMWE.
CSeq: 1 ACK
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:10.200.33.126:5066 —>
INVITE sip:[email protected]:5066;transport=UDP SIP/2.0
Via: SIP/2.0/UDP ISP1_Public_IP:5066;branch=z9hG4bK-d8754z-315ac7e2c554f704-1—d8754z-
Max-Forwards: 70
Contact: <sip:[email protected]_Public_IP:5066;transport=UDP>
To: <sip:[email protected]:5066;transport=UDP>
From: “3353”<sip:[email protected]:5066;transport=UDP>;tag=35468a11
Call-ID: M2Q5MDYwMGZjNjYzOThhMzNlMTMxZGI3ZTY0N2RlMWE.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Zoiper for Windows 2.43 r24984
Authorization: Digest username=“3353”,realm=“asterisk”,nonce=“33a37f46”,uri=“sip:[email protected]:5066;transport=UDP”,response=“f6f9835da16e3a02679e2681fff21fb5”,algorithm=MD5
Allow-Events: presence, kpml
Content-Length: 329

v=0
o=Zoiper_user 0 0 IN IP4 ISP1_Public_IP
s=Zoiper_session
c=IN IP4 ISP1_Public_IP
t=0 0
m=audio 8000 RTP/AVP 3 0 8 110 98 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (15 headers 15 lines) —
Sending to 10.200.33.126:5066 (no NAT)
Using INVITE request as basis request - M2Q5MDYwMGZjNjYzOThhMzNlMTMxZGI3ZTY0N2RlMWE.
Found peer ‘3353’ for ‘3353’ from 10.200.33.126:5066
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 110
Found RTP audio format 98
Found RTP audio format 101
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format speex for ID 110
Found audio description format iLBC for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - (g729|ulaw|alaw|gsm|g723), peer - audio=(ulaw|gsm|alaw|ilbc|speex)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port ISP1_Public_IP:8000
Looking for 3331 in from-internal (domain 10.1.0.20)
sip_route_dump: route/path hop: <sip:[email protected]_Public_IP:5066;transport=UDP>

<— Transmitting (no NAT) to 10.200.33.126:5066 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ISP1_Public_IP:5066;branch=z9hG4bK-d8754z-315ac7e2c554f704-1—d8754z-;received=10.200.33.126
From: “3353”<sip:[email protected]:5066;transport=UDP>;tag=35468a11
To: <sip:[email protected]:5066;transport=UDP>
Call-ID: M2Q5MDYwMGZjNjYzOThhMzNlMTMxZGI3ZTY0N2RlMWE.
CSeq: 2 INVITE
Server: FPBX-13.0.195.28(13.28.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]_Public_IP:5066>
Content-Length: 0

<------------>
Audio is at 20870
Adding codec ulaw to SDP
Adding codec g729 to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g723 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.200.33.201:43430:
INVITE sip:[email protected]_Public_IP:43430;rinstance=ac84e9583c6d15d2;transport=UDP SIP/2.0
Via: SIP/2.0/UDP PBX_Public_IP:5066;branch=z9hG4bK6583b52e
Max-Forwards: 70
From: “IT Support” <sip:[email protected]_Public_IP:5066>;tag=as59a14a7d
To: <sip:[email protected]_Public_IP:43430;rinstance=ac84e9583c6d15d2;transport=UDP>
Contact: <sip:[email protected]_Public_IP:5066>
Call-ID: [email protected]_Public_IP:5066
CSeq: 102 INVITE
User-Agent: FPBX-13.0.195.28(13.28.0)
Date: Fri, 20 Sep 2019 15:22:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: “IT Support” <sip:[email protected]_Public_IP>
Content-Type: application/sdp
Content-Length: 392

v=0
o=root 2128058025 2128058025 IN IP4 PBX_Public_IP
s=Asterisk PBX 13.28.0
c=IN IP4 PBX_Public_IP
t=0 0
m=audio 20870 RTP/AVP 0 18 8 3 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv


<— Transmitting (no NAT) to 10.200.33.126:5066 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ISP1_Public_IP:5066;branch=z9hG4bK-d8754z-315ac7e2c554f704-1—d8754z-;received=10.200.33.126
From: “3353”<sip:[email protected]:5066;transport=UDP>;tag=35468a11
To: <sip:[email protected]:5066;transport=UDP>;tag=as4b9c2682
Call-ID: M2Q5MDYwMGZjNjYzOThhMzNlMTMxZGI3ZTY0N2RlMWE.
CSeq: 2 INVITE
Server: FPBX-13.0.195.28(13.28.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]_Public_IP:5066>
P-Asserted-Identity: “Kapil Dev Sharma” <sip:[email protected]>

You need to describe your network setup, in terms of wan, lan, local and/or remote endpoints.

WAN IP: 65.X.X.X:5066
Local IP PBX: 10.1.0.20:5066
RTP packets starts and ends: 20001 - 21000 ( And same are forwarded to PBX)
For data: 10.200.33.0/24
For voice: 10.200.66.0/24
Remote end points: Softconsole (zoiper and 2 deskphones) are using the same network.

Sounds like SIP is not configured properly for the external IP address to me…

If voice lan is 10.200, the local IP for FreePBX should also be 10.200

Also, RTP by default goes from 10000 - 20000

There were some entries missing in the Ip tables.

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