vicidial.org

I'm having these 3 issues after a fresh iciBox_v8_1.x86_64-8.1.2 install:

1) Asterisk is not starting on boot. If I start manually it works.

2)I keep getting this on CLI:

sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data

3)No sound when logging in using WebRTC viciphone.

Here's the CLI:

[Oct 31 10:42:44] == WebSocket connection from '12.226.124.15:56590' for protocol 'sip' accepted using version '13'
[Oct 31 10:42:45] ERROR[4442]: tcptls.c:447 tcptls_stream_close: SSL_shutdown() failed: error:00000005:lib(0):func(0):DH lib, Underlying BIO error: Broken pipe
[Oct 31 10:42:45] == WebSocket connection from '12.226.124.15:56307' forcefully closed due to fatal write error
[Oct 31 10:42:45] -- Registered SIP '2525' at 12.226.124.15:56590

I do not hear the initial "You are the only......"

Issue seems to be with the certificate (I've used vicibox-certbot). I did not get any errors while installation.

ERROR[4675]: tcptls.c:447 tcptls_stream_close: SSL_shutdown() failed: error:00000005:lib(0):func(0):DH lib, Underlying BIO error: Broken pipe

Any ideas?

I get this immediately after login:

Agent Screen:

No one is in your session: 8600051

WebRTC log shows:

REGISTERED

2018-10-31 13:45:31 =>
displayName: 2525
uri: 2525@66.XXX.X.146
authorizationUser: 2525
password: m5Y4FLgd543543weeWWW
wsServers: wss://dialer.certf.com:8089/ws

CLI:

ERROR[4675]: tcptls.c:447 tcptls_stream_close: SSL_shutdown() failed: error:00000005:lib(0):func(0):DH lib, Underlying BIO error: Broken pipe

Did you enter the location of your certificates in /etc/apache2/vhosts.d/1111-default-ssl.conf

SSLCertificateFile /etc/certbot/live/FQDN/cert.pem
SSLCACertificateFile /etc/certbox/live/FQDN/fullchain.pem
SSLCertificateKeyFile /etc/certbox/live/FQDN/privkey.pem

Now I'm getting this:

viciphone debug log:

2018-10-31 15:21:53 =>
displayName: 2525
uri: 2525@66.58.85.146
authorizationUser: 2525
password: m5Y4FLQOUfROo2weeWWW
wsServers: wss://dialer.crft.com:8089/ws
2018-10-31 15:21:56 => Got Invite from <0000000000> "ACagcW15410136786666666666666666"
2018-10-31 15:21:56 => Auto-Answered Call

Call gets auto-answered on viciphone. But here's the CLI:

[Oct 31 15:12:26] == WebSocket connection from '12.333.333.65:49729' for protocol 'sip' accepted using version '13'
[Oct 31 15:12:26] -- Registered SIP '2525' at 12.XXX.XXX.65:49729
[Oct 31 15:12:26] NOTICE[3405]: chan_sip.c:24639 handle_response_peerpoke: Peer '2525' is now Reachable. (105ms / 2000ms)
[Oct 31 15:12:29] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 31 15:12:29] == DTLS ECDH initialized (automatic), faster PFS enabled
[Oct 31 15:12:29] == Using SIP RTP CoS mark 5
[Oct 31 15:12:29] -- Called 2525
[Oct 31 15:12:29] -- SIP/2525-00000003 is ringing

asterisk is not detecting the answer.

The debug dates 15:21

Asterisk dates 15:12

[Oct 31 20:43:08] == WebSocket connection from '12.226.XX.65:57378' for protocol 'sip' accepted using version '13'
[Oct 31 20:43:08] -- Registered SIP '2525' at 12.226.XX.65:57378
[Oct 31 20:43:08] > Saved useragent "VICIphone 1.0-rc1" for peer 2525
[Oct 31 20:43:08] NOTICE[22371]: chan_sip.c:24639 handle_response_peerpoke: Peer '2525' is now Reachable. (100ms / 2000ms)
[Oct 31 20:43:11] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 31 20:43:11] == DTLS ECDH initialized (automatic), faster PFS enabled
[Oct 31 20:43:11] == Using SIP RTP CoS mark 5
[Oct 31 20:43:11] -- Called 2525
[Oct 31 20:43:11] -- SIP/2525-00000000 is ringing

WebRTC log:

2018-10-31 20:43:44 =>
displayName: 2525
uri: 2525@66.XXX.4.146
authorizationUser: 2525
password: m5Y4FLQOUfRygyg666gg
wsServers: wss://dialer.clrt.com:8089/ws
2018-10-31 20:43:47 => Got Invite from <0000000000> "ACagcW15410329896666666666666666"
2018-10-31 20:43:52 => Answered Call

if vicibox-certbot ran without any real issues then it should have set the correct SSL certificates in /etc/asterisk/http.conf and /etc/apache2/vhosts.d/1111-default-ssl.conf.

The only thing you really need to do is edit the 'WebRTC' template to point to the certificates as well. So I would double check that and see if that's your issue. You can find it under Admin --> Templates. It looks like you already got the WSS set correctly since it's registering.

Other then that I'm not sure. Some of the issues sound network related. All 3 issues can be attributed to network issues but that doesn't mean it's their cause. There is an init script log at /var/log/vicidial.log that should show you what the init script was seeing during start up. Item 2 and 3 almost 100% describe a NAT traversal issue.

I'm also getting this error on the asterisk console while testing vicibox 8.1.2:

Serious Network Trouble; __sip_xmit returns error for pkt data

It happens after a while when you are logged in as agent with webphone, not immediately.

It seems to be WebRTC related:
https://community.freepbx.org/t/chan-si ... a/39534/16

I also found this:

In chan_sip, a “__sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error” error typically means that there’s a misconfiguration on the IP addressing on your system. Asterisk is trying to send a SIP message, and passes the network packet to the operating system, and the operating system says “Whoa there buddy, that’s not right”.

However i have checked all the obvious places but my IP is correct. Also ran the update IP script.

Another find on the FreePBX forum:

So since I posted the OP, I found out what the issue was. (or at least eliminated it by changing a setting.) On old versions of FreePBX, in Advanced SIP settings, in order to enable TCP connections, you had to manually type:

tcpenable =yes
In new versions, there is now a button to enable TCP. Having both settings enabled seems to cause these issues. Most likely, the config file for sip.conf gets messed up having these similar/identical settings. Removing the old entry eliminates the errors.

This is a Vicibox 8.1.2 server with SVN 5053.
The cert was installed with vicibox-certbot.
I checked /etc/asterisk/http.conf and /etc/apache2/vhosts.d/1111-default-ssl.conf and both have the correct paths.
The webphone is working normally there are no functional problems as far as i can see.

I also see this error:

ERROR[4675]: tcptls.c:447 tcptls_stream_close: SSL_shutdown() failed: error:00000005:lib(0):func(0):DH lib, Underlying BIO error: Broken pipe

dennis,

you ever get the broken pipe error situated?

Nox

Not actively, but i don't think i see them anymore or i have become blind for them

I do keep seeing 'Serious Network Trouble' and 'SRTCP unprotect failed because of authentication failure'

that just started recently right? does it effect the audio on some agents calls only?

I just saw the Underlying BIO error too today. We have no audio problems resulting from any of them. Also the vicidial group said all these warnings can be safely ignored.

sounds good