Incoming call doesn't work, outgoing call yes

bitter · October 21, 2022, 6:22pm

Hi all,
I am new in pbx world so apologise me if I don’t explain myself very well.
I have setup a vital pbx in high availability for test purposes (version 3.2.3-5, asterisk 18.12.1), and configured some extensions with softphone (ext 101,102,103). Everything works like a charm, calls from outside to inside and viceversa works.
Then i decided to swap a softphone ext 102 with a Fanvil XS3, registrations to pbx is ok but it can’t receive any calls, neither from other extension neither from outside. The call goes in timeout after 4 seconds. Outgoing calls instead work. Already reset it to factory settings but didn’t help. It is very strange I don’t understand how to figure out. There is no firewall between pbx and phones, everything is in the same network. What can I do check?

bitter · October 21, 2022, 8:32pm

Here is an extract from debug:

pbx1*CLI> 
Reliably Transmitting (no NAT) to 109.233.129.13:5061:
OPTIONS sip:sip.messagenet.it SIP/2.0

Via: SIP/2.0/UDP 192.168.178.31:5060;branch=z9hG4bK227b9a27

Max-Forwards: 70

From: "asterisk" <sip:5434980121@192.168.178.31>;tag=as28e1c286

To: <sip:sip.messagenet.it>

Contact: <sip:5434980121@192.168.178.31:5060>

Call-ID: 1421275e3e0401240175c3b21ad98dd0@192.168.178.31:5060

CSeq: 102 OPTIONS

User-Agent: VitalPBX

Date: Fri, 21 Oct 2022 20:15:41 GMT

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

Supported: replaces, timer

Content-Length: 0




---

pbx1*CLI> 

<--- SIP read from UDP:109.233.129.13:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.178.31:5060;branch=z9hG4bK227b9a27;rport=5060;received=87.234.23.2
From: "asterisk" <sip:5434980121@192.168.178.31>;tag=as28e1c286
To: <sip:sip.messagenet.it>;tag=83b731ea89efef3e825665e07e69ddd2.782c
Call-ID: 1421275e3e0401240175c3b21ad98dd0@192.168.178.31:5060
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding: 
Accept-Language: en
Supported: 
Server: sip.messagenet.it SIP Proxy
Content-Length: 0
Warning: 392 109.233.129.13:5061 "Noisy feedback tells: pid=20652 req_src_ip=87.234.23.2 req_src_port=5060 in_uri=sip:sip.messagenet.it out_uri=sip:sip.messagenet.it via_cnt==1"

<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '1421275e3e0401240175c3b21ad98dd0@192.168.178.31:5060' Method: OPTIONS

pbx1*CLI> 
Retransmitting #5 (no NAT) to 192.168.178.10:5060:
INVITE sip:102@192.168.178.10:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.178.31:5060;branch=z9hG4bK130f9597

Max-Forwards: 70

From: "+39028773652" <sip:+39028773652@192.168.178.31>;tag=as2cbd2f02

To: <sip:102@192.168.178.10:5060>

Contact: <sip:+39028773652@192.168.178.31:5060>

Call-ID: 23d07da24f2b9bd67674afd2311cccde@192.168.178.31:5060

CSeq: 102 INVITE

User-Agent: VitalPBX

Date: Fri, 21 Oct 2022 20:15:38 GMT

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

Supported: replaces, timer

QUEUE-NAME: Q9001

APPLICATION: Q9001

CALL-TYPE: IN

Remote-Party-ID: "+39028773652" <sip:+39028773652@192.168.178.31>;party=calling;privacy=off;screen=no

Content-Type: application/sdp

Content-Length: 327



v=0

o=root 2025009406 2025009406 IN IP4 192.168.178.31

s=Asterisk PBX 18.12.1

c=IN IP4 192.168.178.31

t=0 0

m=audio 13700 RTP/AVP 18 8 0 101

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=maxptime:150

a=sendrecv


---

pbx1*CLI> 
Retransmitting #6 (no NAT) to 192.168.178.10:5060:
INVITE sip:102@192.168.178.10:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.178.31:5060;branch=z9hG4bK130f9597

Max-Forwards: 70

From: "+39028773652" <sip:+39028773652@192.168.178.31>;tag=as2cbd2f02

To: <sip:102@192.168.178.10:5060>

Contact: <sip:+39028773652@192.168.178.31:5060>

Call-ID: 23d07da24f2b9bd67674afd2311cccde@192.168.178.31:5060

CSeq: 102 INVITE

User-Agent: VitalPBX

Date: Fri, 21 Oct 2022 20:15:38 GMT

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

Supported: replaces, timer

QUEUE-NAME: Q9001

APPLICATION: Q9001

CALL-TYPE: IN

Remote-Party-ID: "+39028773652" <sip:+39028773652@192.168.178.31>;party=calling;privacy=off;screen=no

Content-Type: application/sdp

Content-Length: 327



v=0

o=root 2025009406 2025009406 IN IP4 192.168.178.31

s=Asterisk PBX 18.12.1

c=IN IP4 192.168.178.31

t=0 0

m=audio 13700 RTP/AVP 18 8 0 101

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=maxptime:150

a=sendrecv


---

pbx1*CLI> 
[2022-10-21 22:15:44] WARNING[2216]: chan_sip.c:4151 retrans_pkt: Retransmission timeout reached on transmission 23d07da24f2b9bd67674afd2311cccde@192.168.178.31:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[2022-10-21 22:15:44] WARNING[2216]: chan_sip.c:4175 retrans_pkt: Hanging up call 23d07da24f2b9bd67674afd2311cccde@192.168.178.31:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

pbx1*CLI> 
 Extension Hangup

pbx1*CLI> 
 Extension Hangup - Incoming Call

eevans · October 22, 2022, 6:54am

I have the same issue. Did you get it resolved?

mo10 · October 22, 2022, 7:08am

Really with high availability as well?

Did you try to go back to that softphone? Does it work then again?

You check with sngrep as well what is happening.

miguel · October 22, 2022, 8:50am

What technology are you using for your trunk? PJSIP or SIP?

Did you configure in the bind address the floating IP?

bitter · October 22, 2022, 9:00am

Trying to go back to softphone it works again.
I have only done some debug trying to find the problem.

bitter · October 22, 2022, 9:01am

Tried either floating IP either the pbx’s IP address in active.
I have tried only with SIP

miguel · October 22, 2022, 9:16am

My advice, use PJSIP instead of SIP. PJSIP use the SIP protocol, so you don’t have to be afraid about your provider compatibility.

Finally, define the floating IP as the bind address in the PJSIP settings.

eevans · October 22, 2022, 9:17am

O use sip wtih callcentric providing the turnks. They are looking at it as well…can call out no problem, but calls are not routed properly coming in and both they and I think the config looks good. I even tried turning the firewall off just as a test…no luck, and turned it back on.
Phones provision properly and all works, just cannot get calls.

miguel · October 22, 2022, 9:36am

Keep in mind that in the latest version of VitalPBX PSIP runs over port 5060/UDP by default, and SIP over port 5062/UDP.

So, if you created your trunk as a SIP trunk, then it’s possible that your call is being rejected by the PBX because is reaching the port 5060 instead of 5062.

bitter · October 22, 2022, 2:29pm

of course, I have setup pjsip port 5070-5072 and used 5060 for sip. I have tried also extension pjsip with same result.
How can I share here my sngrep report?

bitter · October 22, 2022, 2:58pm

I destroyed cluster and finally phone got worked. I think there is something broken in ha system. I think I will not use for a while… this easy problem made me stupid