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Able to make outbound calls, inbound mix in with Callcentric

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Harry Torres

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Intermediate Cert.
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The title may have not made sense...trying to stay away from saying "its not working".

Setting up a HyperV Debian 9 VM guest with 3CX. Setup new phones, outbound call works. I'm testing inbound with just a voicemail box at this point and it does not work right. With Callcentric, it seems as if Callcentric's error message AND the PBX answer at the same time, and with Flowroute, I just get an error that the number is not valid...

Both SIP trunks are setup with inbound rules, and firewall ports are opened to that IP.

Thoughts?

How much of the log can I post on here?

Thanks.
 
If you see a 3CX Activity Log, of an incoming call, that fails, unless you have changed it to display all of the details, it should not be that long. You can "X" out your Public IP, or URL.

If you don't get a log of calls from one trunk or the other, I'd contact that provider for assistance in determining why the call did not go through (show in the log).
 
Pasting below..I Xed out ph numbers, names, and external IP...

Code:
10/15/2018 6:04:13 PM - Leg L:4.2[VMail:999] is terminated: Cause: BYE from local
10/15/2018 6:04:13 PM - L:4.2[VMail:999] got Terminated Send Req BYE from 0.0.0.0:0 tid=0afe494668bdfa6b Call-ID=zllyGVwJwrXUIFYUFR92gA..:

BYE sip:[email protected]:5483;rinstance=770ffa8b338041f5 SIP/2.0

Via: SIP/2.0/ ;branch=z9hG4bK-524287-1---0afe494668bdfa6b;rport

Max-Forwards: 70

Contact: <sip:[email protected]:5060>

To: <sip:[email protected]:5060>;tag=dd7d5513

From: "xxxxxxx1612"<sip:[email protected]:5060;nf=e>;tag=edac550e

Call-ID: zllyGVwJwrXUIFYUFR92gA..

CSeq: 2 BYE

User-Agent: 3CXPhoneSystem 15.5.15502.6 (15502)

Content-Length: 0



10/15/2018 6:04:13 PM - Terminated from <sip:[email protected]:5060>;tag=dd7d5513 to "xxxxxxx1612"<sip:[email protected]:5060;nf=e>;tag=edac550e; reason: LocalBye
10/15/2018 6:04:13 PM - L:4.2[VMail:999] Sending: OnSendReq Send Req BYE from 0.0.0.0:0 tid=e62fb97a3dff4565 Call-ID=zllyGVwJwrXUIFYUFR92gA..:

BYE sip:[email protected]:5483;rinstance=770ffa8b338041f5 SIP/2.0

Via: SIP/2.0/ ;branch=z9hG4bK-524287-1---e62fb97a3dff4565;rport

Max-Forwards: 70

Contact: <sip:[email protected]:5060>

To: <sip:[email protected]:5060>;tag=dd7d5513

From: "xxxxxxx1612"<sip:[email protected]:5060;nf=e>;tag=edac550e

Call-ID: zllyGVwJwrXUIFYUFR92gA..

CSeq: 2 BYE

Content-Length: 0



10/15/2018 6:04:13 PM - SendMsg from "xxxxxxx1612"<sip:[email protected]:5060;nf=e>;tag=edac550e to <sip:[email protected]:5060>;tag=dd7d5513
10/15/2018 6:04:13 PM - Leg L:4.1[Line:10000<<xxxxxxx1612] is terminated: Cause: BYE from local
10/15/2018 6:04:13 PM - L:4.1[Line:10000<<xxxxxxx1612] got Terminated Send Req BYE from 0.0.0.0:0 tid=85caad01afd0bd49 [email protected]:

BYE sip:[email protected]:5060;transport=udp SIP/2.0

Via: SIP/2.0/ ;branch=z9hG4bK-524287-1---85caad01afd0bd49;rport

Max-Forwards: 70

Contact: <sip:[email protected]:5060>

To: "xxxxxxx" <sip:[email protected]>;tag=3748629820-861047

From: <sip:[email protected]>;tag=037bd46c

Call-ID: [email protected]

CSeq: 2 BYE

User-Agent: 3CXPhoneSystem 15.5.15502.6 (15502)

Content-Length: 0



10/15/2018 6:04:13 PM - Terminated from "xxxxxxx" <sip:[email protected]>;tag=3748629820-861047 to <sip:[email protected]>;tag=037bd46c; reason: LocalBye
10/15/2018 6:04:13 PM - L:4.1[Line:10000<<xxxxxxx1612] Sending: OnSendReq Send Req BYE from 0.0.0.0:0 tid=ed7eee38b9c9810e [email protected]:

BYE sip:[email protected]:5060;transport=udp SIP/2.0

Via: SIP/2.0/ ;branch=z9hG4bK-524287-1---ed7eee38b9c9810e;rport

Max-Forwards: 70

Contact: <sip:[email protected]:5060>

To: "xxxxxxx" <sip:[email protected]>;tag=3748629820-861047

From: <sip:[email protected]>;tag=037bd46c

Call-ID: [email protected]

CSeq: 2 BYE

Content-Length: 0



10/15/2018 6:04:13 PM - SendMsg from <sip:[email protected]>;tag=037bd46c to "xxxxxx" <sip:[email protected]>;tag=3748629820-861047
10/15/2018 6:04:13 PM - [CM503008]: Call(C:4): Call is terminated
10/15/2018 6:04:13 PM - Blocking refers for the Call(C:4)
10/15/2018 6:04:13 PM - Call(C:4) is terminated
10/15/2018 6:04:13 PM - [CM503021]: Call(C:4): ACK is not received from sip:[email protected]
10/15/2018 6:04:13 PM - ACK not received from "xxxxxxx" <sip:[email protected]>;tag=3748629820-861047 to <sip:[email protected]>;tag=037bd46c
10/15/2018 6:04:13 PM - L:4.1[Line:10000<<14842941612] Sending: OnSendResp Send 200/INVITE from 0.0.0.0:0 tid=-e23e338e2b6da7684c9f2d8cb8ee0013 [email protected]:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 204.11.192.162:5060;branch=z9hG4bK-e23e338e2b6da7684c9f2d8cb8ee0013

Contact: <sip:[email protected]:5060>

To: <sip:[email protected]>;tag=037bd46c

From: "xxxxxxx" <sip:[email protected]>;tag=3748629820-861047

Call-ID: [email protected]

CSeq: 1 INVITE

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

Content-Type: application/sdp

Supported: replaces, timer

User-Agent: 3CXPhoneSystem 15.5.15502.6 (15502)

Content-Length: 309



v=0

o=3cxPS 29942881762410496 22780006996377601 IN IP4 XXX.XXX.XXX.XXX

s=3cxPS Audio call

c=IN IP4 XXX.XXX.XXX.XXX

t=0 0

m=audio 9006 RTP/AVP 0 8 18 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

10/15/2018 6:04:13 PM - SendMsg from <sip:[email protected]>;tag=037bd46c to "call id removed" <sip:[email protected]>;tag=3748629820-861047
10/15/2018 6:04:12 PM - CSTA: onReadyToSend: CSTA Send 200/INFO from 0.0.0.0:0 tid=63628962590011717 Call-ID=Z_jLxUFJlsqoX07JMoWfNA..:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.0.215:5060;branch=z9hG4bK63628962590011717

Contact: <sip:108>

To: <sip:[email protected]>;tag=525ffe6e

From: <sip:[email protected]:5060>;tag=1486839717

Call-ID: Z_jLxUFJlsqoX07JMoWfNA..

CSeq: 8 INFO

User-Agent: 3CXPhoneSystem 15.5.15502.6 (15502)

Content-Length: 0
 
Eh a number snuck in...Im not concerned with it lol.
 
10/15/2018 6:04:13 PM - [CM503021]: Call(C:4): ACK is not received from sip:[email protected]

This (ACK not received) may indicate that not all ports are open. Does the 3CX Firewall Checker all pass? Where are you attempting to send incoming calls? Have You tried simplifying things and just sending them to one extension, as a test.
 
Firewall test passes...I did simplify it by having calls go to an extension without a phone...I just want the voicemail prompt to succeed...thats how Im testing it after I tried making inbound calls to phones earlier...what would happen earlier is the phones would ring, and when I hung up my cell phone (the party calling in) the desk phones kept ringing...I had to reject the calls or go into the active calls in the admin console and remove them.

I should also mention that there is another PBX on site, by another provider...keep in mind that the outbound 3CX calls worked beautifully, but can their system somehow be disrupting SIP traffic inbound? I don't know much about it and I don't have access...but according to my boss, and I will double and triple check with him, this customer has fios and it goes from the verizon router, to a box that belongs to the phone provider in question, then supposedly a fully open connection to an ASUS router, where I opened ports at.

I'm not on site at the moment, but I can test it as I have remote access to the admin console...I won't be on site until later in the week...

If it turns out that their system is causing this issue, I will have to cloud host 3CX until the numbers are ported and that hardware is out of there...I need this on premise though as they have a fax machine...
 
This mean anything? Getting these now...

Code:
10/15/2018 6:59:18 PM - Exception: SipMessage::Exception Missing header Contact @ SipMessage.cxx:1393
10/15/2018 6:59:01 PM - Exception: ParseException ParseBuffer.cxx:87, Parse failed Expected " " in context: Contents Messages-Waiting: no Message-Account: sip:[email protected] Voice-Message: 0/0 (0/0) Fax-Message: 0/0 (0/0) None: 0/0 (0/0) ^ @ ParseBuffer.cxx:87
10/15/2018 6:57:41 PM - Exception: SipMessage::Exception Missing header Contact @ SipMessage.cxx:1393
10/15/2018 6:55:41 PM - Exception: SipMessage::Exception Missing header Contact @ SipMessage.cxx:1393
10/15/2018 6:54:32 PM - Exception: SipMessage::Exception Missing header Contact @ SipMessage.cxx:1393
10/15/2018 6:52:49 PM - Exception: SipMessage::Exception Missing header Contact @ SipMessage.cxx:1393
10/15/2018 6:51:49 PM - Exception: SipMessage::Exception Missing header Contact @ SipMessage.cxx:1393
10/15/2018 6:49:55 PM - Exception: SipMessage::Exception Missing header Contact @ SipMessage.cxx:1393
10/15/2018 6:48:55 PM - Exception: SipMessage::Exception Missing header Contact @ SipMessage.cxx:1393
10/15/2018 6:48:15 PM - Exception: SipMessage::Exception Missing header Contact @ SipMessage.cxx:1393
10/15/2018 6:47:11 PM - Exception: SipMessage::Exception Missing header Contact @ SipMessage.cxx:1393
10/15/2018 6:46:11 PM - Exception: SipMessage::Exception Missing header Contact @ SipMessage.cxx:1393
10/15/2018 6:45:09 PM - Exception: ParseException ParseBuffer.cxx:87, Parse failed Expected " " in context: Contents Messages-Waiting: no Message-Account: sip:[email protected] Voice-Message: 0/0 (0/0) Fax-Message: 0/0 (0/0) None: 0/0 (0/0) ^ @ ParseBuffer.cxx:87
 
I should also mention that there is another PBX on site, by another provider...

Are they "sharing" the same router/ public IP? If so, then, yes, that is probably going to cause a problem. 3CX should have it's own public IP. In many cases, outgoing calls may succeed, but getting a call to properly route in, may be an issue. Things are not going to work well if there are shared ports. It's bad enough (sometimes) when dealing with two, or more sets behind a router, two VoIP Pbx's will be a nightmare keeping the ports straight.

Messages-Waiting: no Message-Account: sip:[email protected]

Offhand this would appear to be some sort of message waiting message from the provider. Does the account have voicemail?
 
They aren't sharing the same public IP, or internal network either, as the current phones don't have an IP on that subnet, but the computers that pass through them do...that may be the issue...I changed the internal IP, no phones came through the multicast, and a call succeeded to an internal voicemail. I managed to factory reset all the phones so the multicast took place again (I changed the IP back to what it was in the end) but as soon as the phones started appearing, mind you no calls are being routed to them, it happened again.

The present system is not sharing but is sharing...to some degree..plus like I said, their equipment is in front of all internet traffic.
 
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