Dead air calling voice mail from phone to freepbx

lortech · March 14, 2016, 6:34am

Phone is registered and using very basic switch/routers like a linksys or d-link and when calling voicemail get dead air. Get this error on the screen. Setting change in freepbx?

WARNING[2180]: chan_sip.c:4024 retrans_pkt: Retransmission timeout reached on transmission [email protected] for seqno 2 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[2016-03-13 23:19:40] WARNING[2180]: chan_sip.c:4053 retrans_pkt: Hanging up call [email protected] - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

james · March 14, 2016, 8:16am

It looks network issue there.

cynjut · March 14, 2016, 4:13pm

Tell us more about your network. This screams NAT error, but without knowing more about your network, it’s really impossible to troubleshoot.

lortech · March 14, 2016, 6:46pm

It is really simple. The dell server is on one port on a linksys router/switch and phone on the other port. I am able to cross ping the laptop to server. Laoptop to phone and server to phone. The line is registered. This is a existing running pbx system that was working but was having intermittent connectivity issues. The swithc GW is 192.168.1.1 server is 192.168.1.2 phone is 192.168.1. Line 1 on the VVx310 server ip is configured to 192.168.1.2. I may have to buy a switch for current and future troubleshooting purposes. It would be a quick check to see if its a nat issue.

lortech · March 14, 2016, 8:35pm

Okay I can now reach my voice mail!!

I had to look at the rtp settings for sip and they were correct. The Local net however was set to 192.168.0.0 So I changed it to 192.168.1.0 and now can reach voice mail. This however fix the outgoing call issue with FPBX. I can call into the pbx and the polycom vvx310 rings. Calling out, get dead air then a alison message “the party is not answering”

sip set debug on
SIP Debugging enabled

<— SIP read from UDP:192.168.1.3:5060 —>
INVITE sip:[email protected]:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK87777e9d20374D4
From: “200” sip:[email protected];tag=CD5B29A9-4F19F558
To: sip:[email protected];user=phone
CSeq: 1 INVITE
Call-ID: [email protected]
Contact: sip:[email protected]:5060
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomVVX-VVX_310-UA/4.1.6.5374
Accept-Language: en
Supported: 100rel,replaces
Allow-Events: conference,talk,hold
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 337

v=0
o=- 5966 5966 IN IP4 192.168.1.3
s=Polycom IP Phone
c=IN IP4 192.168.1.3
t=0 0
a=sendrecv
m=audio 10024 RTP/AVP 9 102 0 8 18 127
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:127 telephone-event/8000
<------------->
— (15 headers 15 lines) —
Sending to 192.168.1.3:5060 (NAT)
Sending to 192.168.1.3:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘200’ for ‘200’ from 192.168.1.3:5060

<— Reliably Transmitting (NAT) to 192.168.1.3:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK87777e9d20374D4;received=192.168.1.3;rport=5060
From: “200” sip:[email protected];tag=CD5B29A9-4F19F558
To: sip:[email protected];user=phone;tag=as68ee6bb2
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-12.0.74(11.18.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="4336beb4"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:192.168.1.3:5060 —>
ACK sip:[email protected]:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK87777e9d20374D4
From: “200” sip:[email protected];tag=CD5B29A9-4F19F558
To: sip:[email protected];user=phone;tag=as68ee6bb2
CSeq: 1 ACK
Call-ID: [email protected]
Contact: sip:[email protected]:5060
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomVVX-VVX_310-UA/4.1.6.5374
Accept-Language: en
Max-Forwards: 70
Content-Length: 0

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

<— SIP read from UDP:192.168.1.3:5060 —>
INVITE sip:[email protected]:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bKb228bf2c731DBB9
From: “200” sip:[email protected];tag=CD5B29A9-4F19F558
To: sip:[email protected];user=phone
CSeq: 2 INVITE
Call-ID: [email protected]
Contact: sip:[email protected]:5060
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomVVX-VVX_310-UA/4.1.6.5374
Accept-Language: en
Supported: 100rel,replaces
Allow-Events: conference,talk,hold
Authorization: Digest username=“200”, realm=“asterisk”, nonce=“4336beb4”, uri=“sip:[email protected]:5060;user=phone”, response=“643a03acd8b5a7ff581ad1f2d0a41d43”, algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 337

v=0
o=- 5966 5966 IN IP4 192.168.1.3
s=Polycom IP Phone
c=IN IP4 192.168.1.3
t=0 0
a=sendrecv
m=audio 10024 RTP/AVP 9 102 0 8 18 127
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:127 telephone-event/8000
<------------->
— (16 headers 15 lines) —
Sending to 192.168.1.3:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘200’ for ‘200’ from 192.168.1.3:5060
Found RTP audio format 9
Found RTP audio format 102
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 127
Found audio description format G722 for ID 9
Found audio description format G7221 for ID 102
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 127
Capabilities: us - (gsm|ulaw|alaw|g726), peer - audio=(ulaw|alaw|g729|g722|siren7)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.3:10024
Looking for 6045551212 in from-internal (domain 192.168.1.2)
list_route: hop: sip:[email protected]:5060

<— Transmitting (NAT) to 192.168.1.3:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bKb228bf2c731DBB9;received=192.168.1.3;rport=5060
From: “200” sip:[email protected];tag=CD5B29A9-4F19F558
To: sip:[email protected];user=phone
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-12.0.74(11.18.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Length: 0

<------------>
Audio is at 17996
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100011 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to x.x.xxx.xx:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 96.xx.xxx.xxx:5060;branch=z9hG4bK1a1664d3
Max-Forwards: 70
From: sip:[email protected];tag=as43096c73
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-12.0.74(11.18.0)
Date: Mon, 14 Mar 2016 20:29:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 316

v=0
o=root 2000331159 2000331159 IN IP4 96.xx.xxx.xxx
s=Asterisk PBX 11.18.0
c=IN IP4 96.xx.xxx.xxx
t=0 0
m=audio 17996 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<— SIP read from UDP:x.x.xxx.xx:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK1a1664d3;received=192.168.0.2
From: sip:[email protected];tag=as43096c73
To: sip:[email protected];tag=as325f0e8f
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="77ea8108"
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Retransmitting #1 (no NAT) to x.x.xxx.xx:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 96.xx.xxx.xxx:5060;branch=z9hG4bK1a1664d3
Max-Forwards: 70
From: sip:[email protected];tag=as43096c73
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-12.0.74(11.18.0)
Date: Mon, 14 Mar 2016 20:29:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 316

v=0
o=root 2000331159 2000331159 IN IP4 96.xx.xxx.xxx
s=Asterisk PBX 11.18.0
c=IN IP4 96.xx.xxx.xxx
t=0 0
m=audio 17996 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<— SIP read from UDP:x.x.xxx.xx:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK1a1664d3;received=192.168.0.2
From: sip:[email protected];tag=as43096c73
To: sip:[email protected];tag=as325f0e8f
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="77ea8108"
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Retransmitting #2 (no NAT) to x.x.xxx.xx:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 96.xx.xxx.xxx:5060;branch=z9hG4bK1a1664d3
Max-Forwards: 70
From: sip:[email protected];tag=as43096c73
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-12.0.74(11.18.0)
Date: Mon, 14 Mar 2016 20:29:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 316

v=0
o=root 2000331159 2000331159 IN IP4 96.xx.xxx.xxx
s=Asterisk PBX 11.18.0
c=IN IP4 96.xx.xxx.xxx
t=0 0
m=audio 17996 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<— SIP read from UDP:x.x.xxx.xx:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK1a1664d3;received=192.168.0.2
From: sip:[email protected];tag=as43096c73
To: sip:[email protected];tag=as325f0e8f
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="77ea8108"
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Retransmitting #3 (no NAT) to x.x.xxx.xx:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 96.xx.xxx.xxx:5060;branch=z9hG4bK1a1664d3
Max-Forwards: 70
From: sip:[email protected];tag=as43096c73
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-12.0.74(11.18.0)
Date: Mon, 14 Mar 2016 20:29:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 316

v=0
o=root 2000331159 2000331159 IN IP4 96.xx.xxx.xxx
s=Asterisk PBX 11.18.0
c=IN IP4 96.xx.xxx.xxx
t=0 0
m=audio 17996 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<— SIP read from UDP:x.x.xxx.xx:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK1a1664d3;received=192.168.0.2
From: sip:[email protected];tag=as43096c73
To: sip:[email protected];tag=as325f0e8f
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="77ea8108"
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Retransmitting #4 (no NAT) to x.x.xxx.xx:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 96.xx.xxx.xxx:5060;branch=z9hG4bK1a1664d3
Max-Forwards: 70
From: sip:[email protected];tag=as43096c73
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-12.0.74(11.18.0)
Date: Mon, 14 Mar 2016 20:29:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 316

v=0
o=root 2000331159 2000331159 IN IP4 96.xx.xxx.xxx
s=Asterisk PBX 11.18.0
c=IN IP4 96.xx.xxx.xxx
t=0 0
m=audio 17996 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Retransmitting #5 (no NAT) to x.x.xxx.xx:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 96.xx.xxx.xxx:5060;branch=z9hG4bK1a1664d3
Max-Forwards: 70
From: sip:[email protected];tag=as43096c73
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-12.0.74(11.18.0)
Date: Mon, 14 Mar 2016 20:29:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 316

v=0
o=root 2000331159 2000331159 IN IP4 96.xx.xxx.xxx
s=Asterisk PBX 11.18.0
c=IN IP4 96.xx.xxx.xxx
t=0 0
m=audio 17996 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

[2016-03-14 13:29:49] NOTICE[2180]: chan_sip.c:15095 sip_reregister: – Re-registration for [email protected]
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to x.x.xxx.xx:5060:
REGISTER sip:inbound1.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 96.xx.xxx.xxx:5060;branch=z9hG4bK6b7143ff;rport
Max-Forwards: 70
From: sip:[email protected];tag=as42d6035a
To: sip:[email protected]
Call-ID: 2f1529517676c04b69a462830131193d@[::1]
CSeq: 146 REGISTER
User-Agent: FPBX-12.0.74(11.18.0)
Authorization: Digest username=“xxxxxxx”, realm=“asterisk”, algorithm=MD5, uri=“sip:inbound1.vitelity.net”, nonce=“057b9afe”, response="c442b965c7b8495e699bceb7246bc7fc"
Expires: 120
Contact: sip:[email protected]:5060
Content-Length: 0

<— SIP read from UDP:x.x.xxx.xx:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK6b7143ff;received=192.168.0.2;rport=5060
From: sip:[email protected];tag=as42d6035a
To: sip:[email protected]
Call-ID: 2f1529517676c04b69a462830131193d@[::1]
CSeq: 146 REGISTER
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

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

<— SIP read from UDP:x.x.xxx.xx:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK6b7143ff;received=192.168.0.2;rport=5060
From: sip:[email protected];tag=as42d6035a
To: sip:[email protected];tag=as4a44de44
Call-ID: 2f1529517676c04b69a462830131193d@[::1]
CSeq: 146 REGISTER
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="26ed5ea3"
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Responding to challenge, registration to domain/host name inbound1.vitelity.net
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to x.x.xxx.xx:5060:
REGISTER sip:inbound1.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 96.xx.xxx.xxx:5060;branch=z9hG4bK659b5024;rport
Max-Forwards: 70
From: sip:[email protected];tag=as42d6035a
To: sip:[email protected]
Call-ID: 2f1529517676c04b69a462830131193d@[::1]
CSeq: 147 REGISTER
User-Agent: FPBX-12.0.74(11.18.0)
Authorization: Digest username=“xxxxxxx”, realm=“asterisk”, algorithm=MD5, uri=“sip:inbound1.vitelity.net”, nonce=“26ed5ea3”, response="c4ccb10f29413f930405ce9b3bbe2bf3"
Expires: 120
Contact: sip:[email protected]:5060
Content-Length: 0

<— SIP read from UDP:x.x.xxx.xx:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK659b5024;received=192.168.0.2;rport=5060
From: sip:[email protected];tag=as42d6035a
To: sip:[email protected]
Call-ID: 2f1529517676c04b69a462830131193d@[::1]
CSeq: 147 REGISTER
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

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

<— SIP read from UDP:x.x.xxx.xx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK659b5024;received=192.168.0.2;rport=5060
From: sip:[email protected];tag=as42d6035a
To: sip:[email protected];tag=as4a44de44
Call-ID: 2f1529517676c04b69a462830131193d@[::1]
CSeq: 147 REGISTER
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Expires: 60
Contact: sip:[email protected]:5060;expires=60
Date: Mon, 14 Mar 2016 20:29:49 GMT
Content-Length: 0

<------------->
— (13 headers 0 lines) —
[2016-03-14 13:29:49] NOTICE[2180]: chan_sip.c:23657 handle_response_register: Outbound Registration: Expiry for inbound1.vitelity.net is 60 sec (Scheduling reregistration in 45 s)
Really destroying SIP dialog ‘2f1529517676c04b69a462830131193d@[::1]’ Method: REGISTER
Retransmitting #6 (no NAT) to x.x.xxx.xx:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 96.xx.xxx.xxx:5060;branch=z9hG4bK1a1664d3
Max-Forwards: 70
From: sip:[email protected];tag=as43096c73
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-12.0.74(11.18.0)
Date: Mon, 14 Mar 2016 20:29:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 316

v=0
o=root 2000331159 2000331159 IN IP4 96.xx.xxx.xxx
s=Asterisk PBX 11.18.0
c=IN IP4 96.xx.xxx.xxx
t=0 0
m=audio 17996 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

[2016-03-14 13:29:57] WARNING[2180]: chan_sip.c:4024 retrans_pkt: Retransmission timeout reached on transmission [email protected]:5060 for seqno 102 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response
[2016-03-14 13:29:57] WARNING[2180]: chan_sip.c:4053 retrans_pkt: Hanging up call [email protected]:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
Scheduling destruction of SIP dialog ‘[email protected]:5060’ in 32000 ms (Method: INVITE)
Audio is at 10162
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (NAT) to 192.168.1.3:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bKb228bf2c731DBB9;received=192.168.1.3;rport=5060
From: “200” sip:[email protected];tag=CD5B29A9-4F19F558
To: sip:[email protected];user=phone;tag=as477be55e
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-12.0.74(11.18.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 258

v=0
o=root 1770520683 1770520683 IN IP4 192.168.1.2
s=Asterisk PBX 11.18.0
c=IN IP4 192.168.1.2
t=0 0
m=audio 10162 RTP/AVP 0 8 127
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:127 telephone-event/8000
a=fmtp:127 0-16
a=ptime:20
a=sendrecv

<------------>
Really destroying SIP dialog ‘[email protected]:5060’ Method: INVITE

<— SIP read from UDP:192.168.1.3:5060 —>
CANCEL sip:[email protected]:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bKb228bf2c731DBB9
From: “200” sip:[email protected];tag=CD5B29A9-4F19F558
To: sip:[email protected];user=phone
CSeq: 2 CANCEL
Call-ID: [email protected]
Contact: sip:[email protected]:5060
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomVVX-VVX_310-UA/4.1.6.5374
Authorization: Digest username=“200”, realm=“asterisk”, nonce=“4336beb4”, uri=“sip:[email protected]:5060;user=phone”, response=“998f232a9254c48c38dfdcac5c77f43a”, algorithm=MD5
Max-Forwards: 70
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 192.168.1.3:5060 (NAT)

<— Reliably Transmitting (NAT) to 192.168.1.3:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bKb228bf2c731DBB9;received=192.168.1.3;rport=5060
From: “200” sip:[email protected];tag=CD5B29A9-4F19F558
To: sip:[email protected];user=phone;tag=as477be55e
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-12.0.74(11.18.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>

<— Transmitting (NAT) to 192.168.1.3:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bKb228bf2c731DBB9;received=192.168.1.3;rport=5060
From: “200” sip:[email protected];tag=CD5B29A9-4F19F558
To: sip:[email protected];user=phone;tag=as477be55e
Call-ID: [email protected]
CSeq: 2 CANCEL
Server: FPBX-12.0.74(11.18.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Retransmitting #1 (NAT) to 192.168.1.3:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bKb228bf2c731DBB9;received=192.168.1.3;rport=5060
From: “200” sip:[email protected];tag=CD5B29A9-4F19F558
To: sip:[email protected];user=phone;tag=as477be55e
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-12.0.74(11.18.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.1.3:5060 —>
ACK sip:[email protected]:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bKb228bf2c731DBB9
From: “200” sip:[email protected];tag=CD5B29A9-4F19F558
To: sip:[email protected];user=phone;tag=as477be55e
CSeq: 2 ACK
Call-ID: [email protected]
Contact: sip:[email protected]:5060
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomVVX-VVX_310-UA/4.1.6.5374
Accept-Language: en
Max-Forwards: 70
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: ACK

<— SIP read from UDP:192.168.1.3:5060 —>
ACK sip:[email protected]:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bKb228bf2c731DBB9
From: “200” sip:[email protected];tag=CD5B29A9-4F19F558
To: sip:[email protected];user=phone;tag=as477be55e
CSeq: 2 ACK
Call-ID: [email protected]
Contact: sip:[email protected]:5060
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomVVX-VVX_310-UA/4.1.6.5374
Accept-Language: en
Max-Forwards: 70
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Reliably Transmitting (NAT) to 192.168.1.3:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK744cff17;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as4f70e8bf
To: sip:[email protected]:5060
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-12.0.74(11.18.0)
Date: Mon, 14 Mar 2016 20:30:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.1.3:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK744cff17;rport
From: “Unknown” sip:[email protected];tag=as4f70e8bf
To: “200” sip:[email protected]:5060;tag=32727B35-E07DBD5C
CSeq: 102 OPTIONS
Call-ID: [email protected]:5060
Contact: sip:[email protected]:5060
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub,sdp-anat
User-Agent: PolycomVVX-VVX_310-UA/4.1.6.5374
Accept-Language: en
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
localhost*CLI>

lortech · March 16, 2016, 12:03am

My pbx is still down. I am tempted to do a full reinstall because there is no one willing to provide some help. I think its the router and will remove it and take the phone and the pbx server to the main dsl router and see if the same problem persist.

If anyone cares to take a look at my config they can but need some info from you before you remote into my desktop.

jarvisswope · March 16, 2016, 1:11am

Is this system, phone all plugged into the same switch

cynjut · March 16, 2016, 6:36pm

Just a quick glance through your mountain of debug output.

Do you know why (or what) is reporting a ‘401’ error when trying to connect?

It looks like you’re having problems with the system authenticating with something.

Another one.

lortech · March 22, 2016, 3:57am

I do not know but its acting erratic. I am backing up the settings and I do not see the back up filed in /var/lib/asterisk/ or var/log/asterisk. Going to install a new distribution and reload the settings on it. I CANNOT find the backed up settings even thou I did back it up!