No calls in or out after upgrade on freepbx

agough · October 18, 2018, 5:36pm

Okay I am new to freepbx and I am really lost. I upgraded all modules to the current stable version and software to the latest current version and now for some reason I have no calls in our out. All settings are still there I have contacted my sip provider and everything looks good on their end so it is something wrong in my box and I don’t know where to look. Any help would be great. Thank you.

PitzKey · October 18, 2018, 5:52pm

“In or out” you mean external calls, correct? How about internal calls (extension to extension) do they work?

If they don’t, are all extensions registered? What do you see under Reports > Asterisk Info > Peers?

Further, could be your network got banned, are you using the built-in firewall?
What do you see under Admin > System Admin > Intrusion Detection?

Finally, what are the logs showing?
https://wiki.freepbx.org/display/SUP/Providing+Great+Debug#ProvidingGreatDebug-AsteriskLogs

agough · October 18, 2018, 5:58pm

Internal to internal works fine. All external calls in and out no longer work. We are not using the built in firewall and we are not blocked in the intrusion detection.

Logs
<— SIP read from UDP:10.10.20.1:5060 —>
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.10.20.1;branch=z9hG4bKb7e1bc2471569A1
From: “Ops” sip:[email protected];tag=74F1043C-D71F828B
To: sip:[email protected];user=phone
CSeq: 1 INVITE
Call-ID: [email protected]
Contact: sip:[email protected]
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.14.0987
Accept-Language: en
Supported: 100rel,replaces
Allow-Events: conference,talk,hold
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 294

v=0
o=- 1539885419 1539885419 IN IP4 10.10.20.1
s=Polycom IP Phone
c=IN IP4 10.10.20.1
t=0 0
a=sendrecv
m=audio 2236 RTP/AVP 117 0 8 18 127
a=rtpmap:117 L16/16000
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
<------------->

e[Kglazierpbx*CLI>
e[0K— (15 headers 13 lines) —
Sending to 10.10.20.1:5060 (NAT)
Sending to 10.10.20.1:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘114’ for ‘114’ from 10.10.20.1:5060

<— Reliably Transmitting (no NAT) to 10.10.20.1:5060 —>
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 10.10.20.1;branch=z9hG4bKb7e1bc2471569A1;received=10.10.20.1

From: “Ops” sip:[email protected];tag=74F1043C-D71F828B

To: sip:[email protected];user=phone;tag=as6cc36d64

Call-ID: [email protected]

CSeq: 1 INVITE

Server: FPBX-14.0.3.19(13.22.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“684f4c50”

Content-Length: 0

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

e[Kglazierpbx*CLI>
e[0K
<— SIP read from UDP:10.10.20.1:5060 —>
ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.10.20.1;branch=z9hG4bKb7e1bc2471569A1
From: “Ops” sip:[email protected];tag=74F1043C-D71F828B
To: sip:[email protected];user=phone;tag=as6cc36d64
CSeq: 1 ACK
Call-ID: [email protected]
Contact: sip:[email protected]
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.14.0987
Accept-Language: en
Max-Forwards: 70
Content-Length: 0

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

e[Kglazierpbx*CLI>
e[0K— (12 headers 0 lines) —

e[Kglazierpbx*CLI>
e[0K
<— SIP read from UDP:10.10.20.1:5060 —>
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.10.20.1;branch=z9hG4bK889a00adB44C4614
From: “Ops” sip:[email protected];tag=74F1043C-D71F828B
To: sip:[email protected];user=phone
CSeq: 2 INVITE
Call-ID: [email protected]
Contact: sip:[email protected]
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.14.0987
Accept-Language: en
Supported: 100rel,replaces
Allow-Events: conference,talk,hold
Authorization: Digest username=“114”, realm=“asterisk”, nonce=“684f4c50”, uri="sip:[email protected];user=phone", response=“8ec37617dc9a48ad75b1a903f7a49688”, algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 294

v=0
o=- 1539885419 1539885419 IN IP4 10.10.20.1
s=Polycom IP Phone
c=IN IP4 10.10.20.1
t=0 0
a=sendrecv
m=audio 2236 RTP/AVP 117 0 8 18 127
a=rtpmap:117 L16/16000
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
<------------->

e[Kglazierpbx*CLI>
e[0K— (16 headers 13 lines) —
Sending to 10.10.20.1:5060 (no NAT)

e[Kglazierpbx*CLI>
e[0KUsing INVITE request as basis request - [email protected]
Found peer ‘114’ for ‘114’ from 10.10.20.1:5060

e[Kglazierpbx*CLI>
e[0KFound RTP audio format 117

e[Kglazierpbx*CLI>
e[0KFound RTP audio format 0

e[Kglazierpbx*CLI>
e[0KFound RTP audio format 8

e[Kglazierpbx*CLI>
e[0KFound RTP audio format 18

e[Kglazierpbx*CLI>
e[0KFound RTP audio format 127
Found audio description format L16 for ID 117
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8

e[Kglazierpbx*CLI>
e[0KPeer audio RTP is at port 10.10.20.1:2236
Looking for 6209371447 in from-internal (domain 10.10.10.3)

e[Kglazierpbx*CLI>
e[0Ksip_route_dump: route/path hop: sip:[email protected]

<— Transmitting (no NAT) to 10.10.20.1:5060 —>
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 10.10.20.1;branch=z9hG4bK889a00adB44C4614;received=10.10.20.1

From: “Ops” sip:[email protected];tag=74F1043C-D71F828B

To: sip:[email protected];user=phone

Call-ID: [email protected]

CSeq: 2 INVITE

Server: FPBX-14.0.3.19(13.22.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Contact: sip:[email protected]:5060

Content-Length: 0

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

e[Kglazierpbx*CLI>
e[0KReliably Transmitting (no NAT) to 10.10.20.1:5060:
OPTIONS sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 10.10.10.3:5060;branch=z9hG4bK3c9f2def

Max-Forwards: 70

From: “Unknown” sip:[email protected];tag=as3e2ce7d4

To: sip:[email protected]

Contact: sip:[email protected]:5060

Call-ID: [email protected]:5060

CSeq: 102 OPTIONS

User-Agent: FPBX-14.0.3.19(13.22.0)

Date: Thu, 18 Oct 2018 17:57:06 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Length: 0

e[Kglazierpbx*CLI>
e[0K
<— SIP read from UDP:10.10.20.1:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.3:5060;branch=z9hG4bK3c9f2def
From: “Unknown” sip:[email protected];tag=as3e2ce7d4
To: “Ops” sip:[email protected];tag=6E0396C0-210BA36F
CSeq: 102 OPTIONS
Call-ID: [email protected]:5060
Contact: sip:[email protected]
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.14.0987
Accept-Language: en
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0

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

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

e[Kglazierpbx*CLI>
e[0K[2018-10-18 11:57:31] e[1;31mWARNINGe[0m[30890]: e[1;37mchan_sip.ce[0m:e[1;37m4068e[0m e[1;37mretrans_pkte[0m: 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 32000ms with no response

e[Kglazierpbx*CLI>
e[0KAudio is at 25762
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec g729 to SDP
Adding codec slin16 to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (no NAT) to 10.10.20.1:5060 —>
SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 10.10.20.1;branch=z9hG4bK889a00adB44C4614;received=10.10.20.1

From: “Ops” sip:[email protected];tag=74F1043C-D71F828B

To: sip:[email protected];user=phone;tag=as4694a4ce

Call-ID: [email protected]

CSeq: 2 INVITE

Server: FPBX-14.0.3.19(13.22.0)

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: 346

v=0

o=root 2072072086 2072072086 IN IP4 10.10.10.3

s=Asterisk PBX 13.22.0

c=IN IP4 10.10.10.3

t=0 0

m=audio 25762 RTP/AVP 0 8 18 117 127

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:117 L16/16000

a=rtpmap:127 telephone-event/8000

a=fmtp:127 0-16

a=ptime:20

a=maxptime:70

a=sendrecv

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

e[Kglazierpbx*CLI>
e[0K
<— Reliably Transmitting (no NAT) to 10.10.20.1:5060 —>
SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 10.10.20.1;branch=z9hG4bK889a00adB44C4614;received=10.10.20.1

From: “Ops” sip:[email protected];tag=74F1043C-D71F828B

To: sip:[email protected];user=phone;tag=as4694a4ce

Call-ID: [email protected]

CSeq: 2 INVITE

Server: FPBX-14.0.3.19(13.22.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Length: 0

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

e[Kglazierpbx*CLI>
e[0K[2018-10-18 11:57:35] e[1;31mWARNINGe[0m[27333][C-0000049a]: e[1;37mchannel.ce[0m:e[1;37m5080e[0m e[1;37mast_prode[0m: Prodding channel ‘SIP/114-000004a0’ failed

e[Kglazierpbx*CLI>
e[0K
<— SIP read from UDP:10.10.20.1:5060 —>
ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.10.20.1;branch=z9hG4bK889a00adB44C4614
From: “Ops” sip:[email protected];tag=74F1043C-D71F828B
To: sip:[email protected];user=phone;tag=as4694a4ce
CSeq: 2 ACK
Call-ID: [email protected]
Contact: sip:[email protected]
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.14.0987
Accept-Language: en
Max-Forwards: 70
Content-Length: 0

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

e[Kglazierpbx*CLI>
e[0K— (12 headers 0 lines) —

e[Kglazierpbx*CLI>
e[0KReally destroying SIP dialog ‘[email protected]’ Method: ACK

e[Kglazierpbx*CLI>

BlazeStudios · October 18, 2018, 9:00pm

That is only the internal side of the call. Not the side that is calling the provider and where your issue might be at. You need that side of the call.

agough · October 18, 2018, 9:59pm

How would I get that?

agough · October 18, 2018, 11:17pm

I have already run tests with my provider for the outside and everything is working. So what more do I need to show you?

agough · October 18, 2018, 11:21pm

Also if I check allow anonymous sip calls we can receive outside calls.

cynjut · October 19, 2018, 2:09pm

Have you set up an “any/any” (no DID, no CID) incoming route? If not, do that and watch your logs for “you should have an inbound route for this DID” messages in your logs.

agough · October 19, 2018, 2:22pm

I do not have a specific any any but do have my main number to any. I can try any any and see what happens.

BlazeStudios · October 19, 2018, 2:25pm

OK, you haven’t been very clear on things during this.

1.) Can you receive inbound calls from your provider?
2.) Can you make outbound calls to your provider?

One post you say you can do either and you say you tested with the provider and it’s all good but you’re still saying you have issues. Please answer those questions with Yes/No only after each. No editorializing.

agough · October 19, 2018, 2:30pm

inbound Yes only if allow anonymous calls checked
outbound No

BlazeStudios · October 19, 2018, 2:39pm

Post your trunk config.

cynjut · October 19, 2018, 2:41pm

Check your logs. The error is in there. It’s probably a misconfiguration in your trunk setup, or you have specified only “DID Specific” inbound routes. The most likely (recently) issue is that your configuration is set up for a specific provider IP address and they are sending the calls from a different IP.

Check your logs. The error is in there. It’s probably a misconfiguration in your trunk setup. The most likely scenario here is that you’ve screwed up your server settings and are trying to send calls to a device that doesn’t accept them, your password is incorrect, or your registration string is screwed up.

/var/log/asterisk/full

agough · October 19, 2018, 2:46pm

OUTBOUND
username=XXXXX-XXXXXXX
type=peer
secret=XXXXXXX
insecure=invite
host=voip.centurylink.com
port=5100
context=from-trunk
transport=UDP
disallow=all
allow=ulaw
directmedia=nat

INBOUND
type=peer

register string [email protected]: XXXXXX:[email protected]:5100/XXXXXXXXXX~3600

BlazeStudios · October 19, 2018, 3:27pm

Well this is wrong. Peers do not do registration, that is not what a peer is.

nat=no

The Inbound section should be completely empty except for the top setting “Username” where that should match what comes after the / in the register string and your register string.

Make the other changes to the outgoing settings.

agough · October 19, 2018, 3:33pm

I have made those changes and still nothing. You need me to post any logs?

BlazeStudios · October 19, 2018, 3:47pm

Yes.

asterisk -rvvvvvvv
sip set debug on

Make the call…get everything from the moment you entered sip set debug on all the way to the end and post it.

agough · October 19, 2018, 3:50pm

<— SIP read from UDP:10.10.20.1:5060 —>
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.10.20.1;branch=z9hG4bK1f48665eC401F0DD
From: “Ops” sip:[email protected];tag=B1997298-12566787
To: sip:[email protected];user=phone
CSeq: 1 INVITE
Call-ID: [email protected]
Contact: sip:[email protected]
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.14.0987
Accept-Language: en
Supported: 100rel,replaces
Allow-Events: conference,talk,hold
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 294

v=0
o=- 1539963713 1539963713 IN IP4 10.10.20.1
s=Polycom IP Phone
c=IN IP4 10.10.20.1
t=0 0
a=sendrecv
m=audio 2222 RTP/AVP 117 0 8 18 127
a=rtpmap:117 L16/16000
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
<------------->

e[Kglazierpbx*CLI>
e[0K— (15 headers 13 lines) —
Sending to 10.10.20.1:5060 (NAT)
Sending to 10.10.20.1:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘114’ for ‘114’ from 10.10.20.1:5060

<— Reliably Transmitting (no NAT) to 10.10.20.1:5060 —>
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 10.10.20.1;branch=z9hG4bK1f48665eC401F0DD;received=10.10.20.1

From: “Ops” sip:[email protected];tag=B1997298-12566787

To: sip:[email protected];user=phone;tag=as53a9b0ef

Call-ID: [email protected]

CSeq: 1 INVITE

Server: FPBX-14.0.3.19(13.22.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“1c10e85c”

Content-Length: 0

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

e[Kglazierpbx*CLI>
e[0K
<— SIP read from UDP:10.10.20.1:5060 —>
ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.10.20.1;branch=z9hG4bK1f48665eC401F0DD
From: “Ops” sip:[email protected];tag=B1997298-12566787
To: sip:[email protected];user=phone;tag=as53a9b0ef
CSeq: 1 ACK
Call-ID: [email protected]
Contact: sip:[email protected]
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.14.0987
Accept-Language: en
Max-Forwards: 70
Content-Length: 0

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

e[Kglazierpbx*CLI>
e[0K— (12 headers 0 lines) —

e[Kglazierpbx*CLI>
e[0K
<— SIP read from UDP:10.10.20.1:5060 —>
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.10.20.1;branch=z9hG4bK7b094069CC07CD70
From: “Ops” sip:[email protected];tag=B1997298-12566787
To: sip:[email protected];user=phone
CSeq: 2 INVITE
Call-ID: [email protected]
Contact: sip:[email protected]
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.14.0987
Accept-Language: en
Supported: 100rel,replaces
Allow-Events: conference,talk,hold
Authorization: Digest username=“114”, realm=“asterisk”, nonce=“1c10e85c”, uri="sip:[email protected];user=phone", response=“d9704960dcf69389418b08f99272a070”, algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 294

v=0
o=- 1539963713 1539963713 IN IP4 10.10.20.1
s=Polycom IP Phone
c=IN IP4 10.10.20.1
t=0 0
a=sendrecv
m=audio 2222 RTP/AVP 117 0 8 18 127
a=rtpmap:117 L16/16000
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
<------------->

e[Kglazierpbx*CLI>
e[0K— (16 headers 13 lines) —
Sending to 10.10.20.1:5060 (no NAT)

e[Kglazierpbx*CLI>
e[0KUsing INVITE request as basis request - [email protected]
Found peer ‘114’ for ‘114’ from 10.10.20.1:5060

e[Kglazierpbx*CLI>
e[0KFound RTP audio format 117

e[Kglazierpbx*CLI>
e[0KFound RTP audio format 0

e[Kglazierpbx*CLI>
e[0Ksip_route_dump: route/path hop: sip:[email protected]

<— Transmitting (no NAT) to 10.10.20.1:5060 —>
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 10.10.20.1;branch=z9hG4bK7b094069CC07CD70;received=10.10.20.1

From: “Ops” sip:[email protected];tag=B1997298-12566787

To: sip:[email protected];user=phone

Call-ID: [email protected]

CSeq: 2 INVITE

Server: FPBX-14.0.3.19(13.22.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Contact: sip:[email protected]:5060

Content-Length: 0

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

e[Kglazierpbx*CLI>
e[0KReliably Transmitting (no NAT) to 10.10.20.1:5060:
OPTIONS sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 10.10.10.3:5060;branch=z9hG4bK44d3f891

Max-Forwards: 70

From: “Unknown” sip:[email protected];tag=as437d6eb3

To: sip:[email protected]

Contact: sip:[email protected]:5060

Call-ID: [email protected]:5060

CSeq: 102 OPTIONS

User-Agent: FPBX-14.0.3.19(13.22.0)

Date: Fri, 19 Oct 2018 15:42:04 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Length: 0

e[Kglazierpbx*CLI>
e[0K
<— SIP read from UDP:10.10.20.1:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.3:5060;branch=z9hG4bK44d3f891
From: “Unknown” sip:[email protected];tag=as437d6eb3
To: “Ops” sip:[email protected];tag=DF34D29C-7BEB59EB
CSeq: 102 OPTIONS
Call-ID: [email protected]:5060
Contact: sip:[email protected]
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.14.0987
Accept-Language: en
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0

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

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

e[Kglazierpbx*CLI>
e[0K[2018-10-19 09:42:24] e[1;31mWARNINGe[0m[30890]: e[1;37mchan_sip.ce[0m:e[1;37m4068e[0m e[1;37mretrans_pkte[0m: 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 32000ms with no response
[2018-10-19 09:42:24] e[1;31mWARNINGe[0m[30890]: e[1;37mchan_sip.ce[0m:e[1;37m4092e[0m e[1;37mretrans_pkte[0m: Hanging up call [email protected]:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

e[Kglazierpbx*CLI>
e[0KAudio is at 13376
Adding codec ulaw to SDP

e[Kglazierpbx*CLI>
e[0KAdding codec alaw to SDP
Adding codec g729 to SDP
Adding codec slin16 to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (no NAT) to 10.10.20.1:5060 —>
SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 10.10.20.1;branch=z9hG4bK7b094069CC07CD70;received=10.10.20.1

From: “Ops” sip:[email protected];tag=B1997298-12566787

To: sip:[email protected];user=phone;tag=as1be1cb7d

Call-ID: [email protected]

CSeq: 2 INVITE

Server: FPBX-14.0.3.19(13.22.0)

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: 346

v=0

o=root 1064739082 1064739082 IN IP4 10.10.10.3

s=Asterisk PBX 13.22.0

c=IN IP4 10.10.10.3

t=0 0

m=audio 13376 RTP/AVP 0 8 18 117 127

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:117 L16/16000

a=rtpmap:127 telephone-event/8000

a=fmtp:127 0-16

a=ptime:20

a=maxptime:70

a=sendrecv

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

e[Kglazierpbx*CLI>
e[0K
<— Reliably Transmitting (no NAT) to 10.10.20.1:5060 —>
SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 10.10.20.1;branch=z9hG4bK7b094069CC07CD70;received=10.10.20.1

From: “Ops” sip:[email protected];tag=B1997298-12566787

To: sip:[email protected];user=phone;tag=as1be1cb7d

Call-ID: [email protected]

CSeq: 2 INVITE

Server: FPBX-14.0.3.19(13.22.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

X-Asterisk-HangupCause: No user responding

X-Asterisk-HangupCauseCode: 18

Content-Length: 0

<------------>
[2018-10-19 09:42:26] e[1;31mWARNINGe[0m[21493][C-00000609]: e[1;37mchannel.ce[0m:e[1;37m5080e[0m e[1;37mast_prode[0m: Prodding channel ‘SIP/114-00000603’ failed

e[Kglazierpbx*CLI>
e[0K
<— SIP read from UDP:10.10.20.1:5060 —>
ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.10.20.1;branch=z9hG4bK7b094069CC07CD70
From: “Ops” sip:[email protected];tag=B1997298-12566787
To: sip:[email protected];user=phone;tag=as1be1cb7d
CSeq: 2 ACK
Call-ID: [email protected]
Contact: sip:[email protected]
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.14.0987
Accept-Language: en
Max-Forwards: 70
Content-Length: 0

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

e[Kglazierpbx*CLI>
e[0K— (12 headers 0 lines) —

e[Kglazierpbx*CLI>
e[0KReally destroying SIP dialog ‘[email protected]’ Method: ACK

e[Kglazierpbx*CLI>

agough · October 19, 2018, 4:13pm

I have a new one with full debug since that one is only targeting a single IP. How do I upload it cause it is way to big to post?

BlazeStudios · October 19, 2018, 4:33pm

Pastebin it and post the link.