Grandstream HT502


(Jeffrey Angus) #1

PBX Version: 15.0.16.61
PBX Distro: 12.7.6-2002-2.sng7
Asterisk Version: 16.9.0

Ext. 3009 PJSIP Fox FAX 5060
Ext. 3031 SIP Fox Payphone 5062

Two questions:

  1. Is there anything special I should do with the extensions?
  2. I need a configuration for the HT502.
    Yes, I’ve done my share of Google.
    All of the “Do this” I’ve found so far doesn’t work.
    Yes, I’m aware the HT502 is a “turd” but it supports rotary (pulse) dialing.
    Ext. 3031 is a Bell System rotary dial pay phone.

Jeff


(Jeffrey Angus) #2

Ok, apparently FX1 is working. (extension 3009)
I can call other extensions with it, and call it from them.
FX2 i

isn’t working. I get a fast busy signal if I trying calling other extensions or calling in.


#3

Change extension 3031 to use pjsip (in advanced settings for the extension, Change SIP Driver). In FXS PORT2 settings, change Primary SIP Server to
192.168.1.32:5060
and set Outbound Proxy blank for both FXS ports. Test; if you have trouble, report whether each port is registered and what happens on attempted inbound and outbound calls.


(Jeffrey Angus) #4

If I read that right, change both ports to 5060.
That did not work.
Focusing on the primary usage, to talk to a rotary dial phone.



This works, both outgoing and incoming call.


#5

For FXS port 2 (presently not registered) starting from the settings shown in your July 10 post, please change Primary SIP Server from
192.168.1.32:5062
to
192.168.1.32:5060
Also, delete the entry for Oubound Proxy (leave the box blank).
Leave local SIP port at 5062.
If you have trouble, report whether it shows registered and if not, what errors (if any) appear in the Asterisk log when it attempts to register.


(Jeffrey Angus) #6

Well, FXS1 is still working, but FXS2 is not.
No dial tone, and calling the extension results in a fast busy signal.



#7

TLS only works if you have valid certification accepted by both ends


(Jeffrey Angus) #8

Thank you.
I changed it to UDP.
That “kind of fixed it.”
I have dial tone on FXS2 now, and can call another extension from it.
However, calling that extension results in fast busy signal.


#9

FirstBaby step accomplished, now take the next one, sngrep might well help that progress.


(Jeffrey Angus) #10

Calling Extension 3009 (FXS2 on the HT502) 192.168.1.182
the PBX is 192.168.1.32
from Extension 2002 (A Digium D4) 192.168.1.235

                                       Call flow for .yADnHIL8FbNcd5NmH.N5QhSCFAcus3f (Color by Request/Response)
                                                        │INVITE sip:3009@192.168.1.32 SIP/2.0
        192.168.1.235:5060            192.168.1.32:5060 │Via: SIP/2.0/UDP 192.168.1.235:5060;rport;branch=z9hG4bKPj45YuSSHaVeoeYvyJY5QRxIV9OHSl-qB8
      ──────────┬─────────          ──────────┬─────────│Max-Forwards: 70
                │        INVITE (SDP)         │         │From: "2002" <sip:2002@192.168.1.32>;tag=tzw6ksnQWrAgcyeobotsrMVuaP5mzBma

18:58:01.898579 │ ──────────────────────────> │ │To: sip:3009@192.168.1.32
+0.053815 │ 401 Unauthorized │ │Contact: “2002” sip:2002@192.168.1.235:5060;ob
18:58:01.952394 │ <────────────────────────── │ │Call-ID: .yADnHIL8FbNcd5NmH.N5QhSCFAcus3f
+0.006961 │ ACK │ │CSeq: 10632 INVITE
18:58:01.959355 │ ──────────────────────────> │ │Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
+0.013295 │ INVITE (SDP) │ │Supported: replaces, 100rel, timer, norefersub
18:58:01.972650 │ ──────────────────────────> │ │Session-Expires: 1800
+0.110529 │ 100 Trying │ │Min-SE: 90
18:58:02.083179 │ <────────────────────────── │ │User-Agent: Digium D40 2_9_8
+0.699697 │ 503 Service Unavailable │ │Content-Type: application/sdp
18:58:02.782876 │ <────────────────────────── │ │Content-Length: 499
│ │ │
│ │ │v=0
│ │ │o=- 290055971 290055971 IN IP4 192.168.1.235
│ │ │s=digphn
│ │ │b=AS:84
│ │ │t=0 0
│ │ │a=X-nat:0
│ │ │m=audio 4006 RTP/AVP 0 8 9 111 18 58 118 58 96
│ │ │c=IN IP4 192.168.1.235
│ │ │b=TIAS:64000
│ │ │a=rtcp:4007 IN IP4 192.168.1.235
│ │ │a=sendrecv
│ │ │a=rtpmap:0 PCMU/8000
│ │ │a=rtpmap:8 PCMA/8000
│ │ │a=rtpmap:9 G722/8000
│ │ │a=rtpmap:111 G726-32/8000
│ │ │a=rtpmap:18 G729/8000
│ │ │a=rtpmap:58 L16/16000
│ │ │a=rtpmap:118 L16/8000
│ │ │a=rtpmap:58 L16-256/16000
│ │ │a=rtpmap:96 telephone-event/8000
│ │ │a=fmtp:96 0-16
│ │ │a=ssrc:56885310 cname:2036aebd36e28212


#11

You need to look at the Session from a higher level, an initial 401 is always expected, what where the next exchanges. sngrep allows you to ‘drill down’ from the initial “session” (you are only interested in INVITES), which would expose all the packet exchanges , further each packet can then be more closely inspected. there is ‘help’ just a ? away


(Jeffrey Angus) #12

True, ? worked, but the help is much like a Unix man page. It assumes you already know what you’re doing and just need to be reminded the magic number (or letter as it were.)
Actually, I’m surprised I got as far as I did with the sngrep command.

I’ll be happy to harvest the information if you tell me how to get it.


#13

Sorry, I can help with fishing lessons but I won’t do your fishing for you. because one ‘you’ easily becomes everyone. However one ‘you’ CAN be an ‘everyman’ and ultimately pay back.

When lost there is always the ‘paid for’ Support, linked to at the top of this page.