SIP Phone Setup Instructions Change Suggestion

App_rpt-users
1

I recently setup SIP on ASL3 so I can connect with zoiper to the node. This works great. I did have to change one thing to get it working, had to add vox. I would think all SIP phones would need to be setup to use vox, but perhaps there is a reason it was not setup this way. If there is no reason, I would recommend changing the extensions.conf to add vox like this:

   [sip-phones]
    exten => 1001,1,Dial(PJSIP/${EXTEN},60,rT)

    exten => ${NODE},1,Ringing
    exten => ${NODE},n,Answer(3000)
    exten => ${NODE},n,Set(NODENUM=${CALLERID(number)})
    exten => ${NODE},n,Playback(extension)
    exten => ${NODE},n,SayDigits(${NODENUM})
    exten => ${NODE},n,Playback(connected)
    exten => ${NODE},n,Playback(rpt/node)
    exten => ${NODE},n,SayDigits(${EXTEN})
    exten => ${NODE},n,rpt(${EXTEN}|Pv)   ; <--- added the 'v' for vox
    exten => ${NODE},n,Hangup
2

True ā€œphoneā€ connections (ā€œPā€ mode) in app_rpt do not need vox and should not use it - they should be using *99 to enable PTT and # to disable PTT. That is why the SIP instructions do not show vox enabled.

3

Ah that does work out of the box. I knew there was something like that but couldnā€™t remember the keys. I think vox is easier, just mute when you donā€™t want to be heard. Less keys to press.

4

Is there any reason ā€˜realā€™ phone connections cannot use vox, other than needing to be careful about keying the radios? Found your github instructions just after posting this morning. Great work.

5

Does anyone want to volunteer to help update the ASL3 documentation? Iā€™m thinking that it would help all to document things like *99 + # for PTT, the ā€˜vā€™ for vox, etc.

6

Itā€™s not that they canā€™t, itā€™s just that they shouldnā€™t in most cases. Just like you wouldnā€™t expect to take an Icom HT out of the box and see VOX enabled by defaultā€¦ most people will not ever use vox. To clarify, by ā€œrealā€ phone, I mean a physical SIP phone on a desk. Of course there are cases where you will use IAX/SIP with app_rpt for other things such as in your case, which is why VOX is available as an option - but it is not the default or the ā€œexpectedā€ use case.

1 Like
7

Yes, I was just digging into the manual on github and wondering if the current autopatch stuff should be replaced completely or merged with Masonā€™s. I am willing to help on this. I added an issue on github and if someone wants to guide me on the best workflow to update mkdocs on github I can do the work as well.

8

It took me a minute to find the issue you added and I moved it to the ASL3 Manual repo where youā€™ll find the ā€œsourceā€ and some general notes on getting started. Happy to help you get started.

9

Ah yes, the readme on that repo is very helpful. I will work from there and let you know if I get stuck.

10

Working on the docs and have thought of some things related to this. I see in rpt.conf where 99 is defined as PTT:

;;;;;; Mandatory Command Codes ;;;;;
1 = ilink,1 ; Disconnect specified link
2 = ilink,2 ; Connect specified link ā€“ monitor only
3 = ilink,3 ; Connect specified link ā€“ tranceive
4 = ilink,4 ; Enter command mode on specified link
70 = ilink,5 ; System status
99 = cop,6 ; PTT (phone mode only)

I can change that to 88 in the config and now you dial *88 for PTT. I feel like I have seen where one can select something other than the * character to prefix commands. Does anyone know if thatā€™s right and where to set it? Also, where does the # to unkey get configured, or is that something built into asterisk? Just curious if one could swap # and * or even configure 8 to key and 9 to unkey if they wanted? Would be good to document if itā€™s possible. I have a friend with a rotary phone connected to an ATA that he can dial his node with.

11

Hi
I am struggling to get zoiper to talk to ASL 3 / pjsip.

Are you able to share your parameters please? Are the zoiper settings the same as they were for chan_SIP?

Thanks & 73

12

I followed these instructions: Setting up a SIP Phone - AllStarLink Manual That should have all the info you need. Not sure about zoiper settings between ASL versions. Host, user, and passwd should get you there.

13

Many thanks for the reply.

Iā€™ve copied that code as carefully as possible but Zoiper just shows error 408.

With PJSIP logger on, nothing appears on the screen, so something is clearly not lined up.

Iā€™ll take a break I think and try again another day!

73

14

408 is a timeout. You probably have a network issue. Make sure you have the right host or ip in zoiper and try adding the port like 192.168.1.100:5060. You might try putting the same info in the optional Outbound Proxy field in Zoiper. Mine connects fine with hostname:port, user, and password in Zoiper and thatā€™s it. Check any firewalls along the way.

2 Likes
15

Thanks for that.

I eventually worked out that chan_pjsip was not loaded (even though it was load = in the modules.conf)

I changed the autoload=yes at the top and it fixed it. Iā€™m sure thereā€™s a better way but I donā€™t know what it is!

1 Like
16

https://docs.asterisk.org/Configuration/Core-Configuration/Configuring-the-Asterisk-Module-Loader/

Just to note that some servers that are tight on resources may not want to use that option.
Pretty much is why we have always used a manual load method.
But systems have plenty more now a days.

17

Ah thanks for that - I will read through and see if I can do it more sensibly.

Much appreciated.
73

18

Just to say I have solved this, having had another look.

The wiki says to add the chan_pjsip stuff at the bottom of modules.conf

In my haste I added it right at the bottom, below [global] . Moving it above that solves the issue. Whoops!

1 Like
19

I will update the doc in the manual to include that, perhaps someone can update the wiki? I donā€™t have an account there yet.

1 Like
20

@G4ZWH Robert, can you post the link in the wiki you got that info from.
I canā€™t seem to find it.