No Inbound and outbound call

Hello

I am testing pascom on: demo.pascom.cloud

I added a trunk sip that looks well registered, but I can not make inbound call and outbound call work.

Here’s how it’s configured:

thanks in advance

Hello,

We were able to register the trunk sip, there was an IP address restriction. So we arrive to make outgoing calls now, but unable to make incoming calls. So this is the same problem that we encounter with the pascom installed on a machine on our LAN. On an incoming call here is the SIP trace that we have:

<— SIP read from UDP:217.15.80.163:5060 —>
INVITE sip:0970752593@192.168.1.109:5060 SIP/2.0
Record-Route: sip:217.15.80.163;lr;ftag=as6bd15184;nat=yes
Via: SIP/2.0/UDP 217.15.80.163:5060;branch=z9hG4bK21b9.73c6aed5.0
Via: SIP/2.0/UDP 95.140.14.245:5060;received=95.140.14.245;branch=z9hG4bK3b9def89;rport=5060
Max-Forwards: 69
From: “0763446464” sip:0763446464@95.140.14.245;tag=as6bd15184
To: sip:0970752593@217.15.80.163:5060
Contact: sip:0763446464@95.140.14.245:5060
Call-ID: 7370e10f1e16fb766fbc4c5b723e4f26@95.140.14.245:5060
CSeq: 102 INVITE
Date: Tue, 12 Dec 2017 14:06:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 430

v=0
o=root 1700249886 1700249886 IN IP4 95.140.14.245
s=Asterisk PBX 11.11.0~dfsg-2+alphalink-1
c=IN IP4 217.15.80.163
t=0 0
m=audio 54022 RTP/AVP 8 0 4 111 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:111 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
— (15 headers 18 lines) —
Sending to 217.15.80.163:5060 (no NAT)
Sending to 217.15.80.163:5060 (no NAT)
Using INVITE request as basis request - 7370e10f1e16fb766fbc4c5b723e4f26@95.140.14.245:5060
Found peer ‘mdc_trunk_conf-1’ for ‘0763446464’ from 217.15.80.163:5060

<— Reliably Transmitting (NAT) to 217.15.80.163:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 217.15.80.163:5060;branch=z9hG4bK21b9.73c6aed5.0;received=217.15.80.163;rport=5060
Via: SIP/2.0/UDP 95.140.14.245:5060;received=95.140.14.245;branch=z9hG4bK3b9def89;rport=5060
From: “0763446464” sip:0763446464@95.140.14.245;tag=as6bd15184
To: sip:0970752593@217.15.80.163:5060;tag=as19b61b81
Call-ID: 7370e10f1e16fb766fbc4c5b723e4f26@95.140.14.245:5060
CSeq: 102 INVITE
Server: Asterisk PBX certified/11.6-cert17
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“pascom”, nonce=“158ea351”
Content-Length: 0

Regards,

Hello,

the invite is directly captured on the (non cloudstack) pbx? The asterisk CLI shows now output all?
In case you see no asterisk CLI output and the registration runs on the same host the invite is recieved from (95.140.14.245) you might need to configure externip and localnet via systemsetings.
In case there is output, this would help.

Best regards,
Steve

Hello,

I admit that I do not understand everything you said.
I did on the PBX in CLI the following command : “sip set debug on” to have these traces on an incoming call

Regards

Hello,

and there were no lines skipped between
Found peer ‘mdc_trunk_conf-1’ for ‘0763446464’ from 217.15.80.163:5060

and

<— Reliably Transmitting (NAT) to 217.15.80.163:5060 —>
SIP/2.0 401 Unauthorized

?

When “sip show registry” shows the registert host you could try to resolve the ip for this host (A or SRV record, depending on trunk) and compare it to the invite source IP. When they differ the asterisk denies the call because this host has not been authenticated yet.

Best regards,
Steve