Some Call drop after 32S

[JeanJean]

Hi,

I have little problem with my 3cx test.

We have a freepbx with SIP Trunk connect to the provider directly with IPVPN.

All call are OK

Same test with 3cx::

Some Call are Ok and other drop after 32Sec.

My provider say it is a 3cx problem. For the sip provider, the problem is 3cx provide no reponse to the reinvite request. Freepbx yes.

Possible that 3cx are not compatible but Freepbx yes ?

thanks for the help

 

[JeanJean]

Hi

thanks for your answer.

The problem is only in Outbound when 3Cx (App or Phone) call a external number.

Exemple: Call Fixed Line Work (For the provider there is no reinvite that for what is Ok)
Call GSM Number: not working because Reinvite is Needed.

All inbound call work.

The trace that i have show is:

Zoiper -> FreePBX -> GSM number: OK

3Cx Webclient -> 3CX-> GSM Number: Not OK

Also tested with Phone/3Cxapp

 

[JeanJean]

 

[NickD_3CX]

I see. 3CX does support re-INVITES, so it could be something in the re-INVITE that is wrong.

Do you have a different provider to try with?
Also you are saying that you are using the WebClient. Does thw same happen when you use a different endpoint, like a mobile app or deskphone?

 

[JeanJean]

Yes same problem with 3Cx APP and desk Phone.

Yes i have test another provider and it is Working.

My porvider don't see the problem because working with Freepbx (Therefore not problem on Trunk Line)

 

[NickD_3CX]

I see your point and how that could be the obvious explanation, but there are skme cases where something that is violating the RFC, and some software 'ignore' it and continue, whereas others may not.

If this is a supported provider, tell me who they are via PM and I will try and replicate this issue and figure it out.
If this is not a supprted provider, I think you should go for one that is. This is the whole reason why we have providers that are supported.

 

[JeanJean]

It is not supported provider.

and not possible to change it

 

[ivan.antic]

JJ,

You have the first INVITE dialog which is fine. 200 OK+SDP sent from provider and you ACK it. Session is established dialog completed.
If something would have went wrong here, the provider would keep pounding you with 200 OK+SDP until it gets an ACK. so you are good.

Why would provider send an re-INVITE back to you?, if the previous dialog is good.
There could be couple or reasons, but in all cases "something" has changed*.

Even in good calls using FreePBX there's an re-INVITE.
So called party is trying to reach out the caller for something, in FreePBX you did sent 200 OK + SDP but not in 3cx.

Is it regular external call?. A calls B and talk with each other.
or A calls B and then B goes back into queue or gets into some crazy things?.

At the end you sent "481 Call/Transaction Does Not Exist" to provider on its re-INVITE.
re-INVITE did never happen from providers perspective.

P.S. We are using Generic VOIP provider setup, and we are not supported providers as well. But all 3cx host are working

 

[JeanJean]

The problem is not only to one number but different number.

 

[JeanJean]

First invite (Starting)

2020/12/28 12:10:38.051646 192.168.0.11:5060 -> PROVIDERIP:5060
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK-524287-1---1e1d9f45d14f0b07;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5060>
From: "xxxxxx"<sip:[email protected]:5060>;tag=8fdd4741
Call-ID: iFSKETmp_i69tyQvkZJUuw..
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 16.0.7.1078 (1078)
Remote-Party-ID: "xxxxxx"<sip:[email protected]:5060>;party=calling
Content-Length: 219

v=0
o=3cxPS 17422623705137152 22359792714514433 IN IP4 192.168.0.11
s=3cxPS Audio call
c=IN IP4 192.168.0.11
t=0 0
m=audio 9106 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

 

[JeanJean]

Reinvite from provider

2020/12/28 12:10:42.516190 PROVIDERIP:5060 -> 192.168.0.11:5060
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP PROVIDERIP:5060;branch=z9hG4bKucbis6006o3mcgk560n0sbk81rns3.1
From: <sip:xxxxxx@PROVIDERIP:5060>;tag=SDfs6b399-1279252891-1609153839080
To: "xxxxxx"<sip:xxxxxx@PROVIDERIP:5060>;tag=8fdd4741
Call-ID: iFSKETmp_i69tyQvkZJUuw..
CSeq: 344292857 INVITE
Contact: <sip:xxxxxx@PROVIDERIP:5060;transport=udp>
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE,NOTIFY
Supported: timer
Accept: application/media_control+xml,application/sdp
Max-Forwards: 68
Content-Length: 212
Content-Type: application/sdp

v=0
o=- 3652208843 3 IN IP4 PROVIDERIP
s=-
c=IN IP4 PROVIDERIP
t=0 0
m=audio 14046 RTP/AVP 8 101
b=AS:29
b=RS:362
b=RR:1087
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

 

[JeanJean]

After this I have the 100 Trying
And provider reply with CANCEL

 

[ivan.antic]

JJ,

Can you change "Registrar/Server/Gateway Hostname or IP" of the trunk to PROVIDERIP instead sip.provider.fr? and re-register the trunk.

I will quote myself:
Is it regular external call?. A calls B and talk with each other.
or A calls B and then B goes back into queue or gets into some crazy things?.

P.S. Make sure you have sip.provider.fr combo in /etc/hosts

FYI: You can use different SIP client against 3cx server. We did tests with Jitsi Deskop for example, when we had troubles with 3cx Web client. But at the end of the day the 3cx client provided most features we needed and it's supported and we like it.

 

[JeanJean]

With the IP registration Failed

 

[JeanJean]

Call A To Call B and speak.

No queue no Third party

 

[ivan.antic]

JJ,

Try with different SIP client against 3cx. I know for a fact that Jitsi Desktop works fine with 3cx.

if you don't find any more clues after this client t-shoot, then there is no other but to call out sip.provider.fr.
Check with them why re-INVITE is happening in a first place, for regular call with already established session and no changes seen in SDP.

I am sure 3x support can't do much as you are with unsupported provider

 

[JeanJean]

Hi thanks for your help.

But doesn't work

For the provider the reinvite is normal in the case you call Mobile Number. But for us the 3cx not reply.

FreePBx yes, therefore for us is 3cx the problem because it is working with other platform.

 

[JeanJean]

The only difference i see is that in the ACK reply from 3cx:

FreePbx: contact is present with other info.
3Cx: not

 

[ivan.antic]

JJ,

can you paste the 200 OK (SDP) and ACK from both FreePBX and 3CX ?

 

[JeanJean]

Thanks for your help

 

[ivan.antic]

JJ,

Now the last peace of puzzle, can you please take sngrep output fom other leg of a call = internal number to 3cx

(With a 3cx web client you will see it as 127.0.0.1:5063 -> 127.0.0.1:5060)

Here's the screen of regular call to external number using 3cx web client (127.0.0.1) to 3cx (150.50.50.250:5060) to provider (10.180.75.10:5060)