Can't connect to Allstar node from SIP phone

AllStarLink
1

Hello All,

I have just updated to the latest allstar version. I am using a Debian 12 system with Cisco SPA525g2 phone. My phone is not connecting to my allstar node.

I have set my extension to 1001 and my node # is 60694.

I have followed these instructions: Setting up a SIP Phone - AllStarLink Manual

I have updated: Update modules.conf

My extensions file:
NODE = 60694 ; change this to your node number
[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(rpt/connected)
exten => ${NODE},n,Playback(rpt/node)
exten => ${NODE},n,SayDigits(${EXTEN})
exten => ${NODE},n,rpt(${EXTEN}|P)
exten => ${NODE},n,Hangup

update pjsip.conf:
[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(rpt/connected)
exten => ${NODE},n,Playback(rpt/node)
exten => ${NODE},n,SayDigits(${EXTEN})
exten => ${NODE},n,rpt(${EXTEN}|P)
exten => ${NODE},n,Hangup

update pjsip.conf:

type=endpoint
transport=transport-udp
context=sip-phones
disallow=all
allow=ulaw
allow=alaw
allow=gsm
auth=1001
aors=1001

callerid=“Allstar-60694-KE4QCM”

type=auth
auth_type=userpass
password=xxxxxxxxx
username=1001

type=aor
max_contacts=4
;contact=sip:6001@192.0.2.1:5060

On my phone everything shows registered ok.

When I dial 1001 from my phone it just rings.
When I dial 60694 from my phone it connects but no audio is heard on my phone.

Does anyone have any suggestions to resolve this issue ?

thanks,

Thomas
KE4QCM

2

Are you in the correct context ?
What is the
context=
for the device in sip.conf ?

It can’t find a match if it is not looking in the context it’s told to.
It’s simpler to keep it all in [radio-control] unless you have good reason not to.
Perhaps it is?

But as stated above, your sip device should be in context=sip-phones

I am not sure why all need for global variables for this.
They seem to create more confusion then they are worth.
It’s much easier understood if you don’t use them like this

exten => 60694,1,Answer
exten => 60694,n,Wait(2)
exten => 60694,n,Playback(rpt/node)
exten => 60694,n,SayDigits(60694)
exten => 60694,n,Rpt(60694|P)

But is only a suggestion. And it is very exacting.

3

…Likely the reason.

4

Hi Mike,

in sips.conf it is configured as
context=public

I changed it to context=sip-phones but it still didn’t work.

-Thomas

5

You might watch asterisk in the foreground and see what the issue is…
And state the message at point where you get error or dialplan stops.

asterisk -rvvv

6

This looks like a stanza for iax client.
Don’t use that for a sip phone connection.
Use my example from above, here again.

exten => 60694,1,Answer
exten => 60694,n,Wait(2)
exten => 60694,n,Playback(rpt/node)
exten => 60694,n,SayDigits(60694)
exten => 60694,n,Rpt(60694|P)

It was familiar to me so I just did not notice at first.

7

What is this suppose to do ? I am confused.

8

Mike,
All my config files for this are made from Setting up a SIP Phone - AllStarLink Manual

Should I be dialing ‘1001’ or ‘60694’ from my phone?

Tom
KE4QCM

9

Well, I do apologize for not being more familiar with pjsip.
Been doing sip for a very long time.

Yes, that is the idea.
If the intention is to call the node from the phone, and be able to control it.

This is the line that makes the connection to the node.

exten => ${NODE},n,rpt(${EXTEN}|P)

And this

does not do that. It is a node you are connecting, not a non radio extension

If I were to rewrite that

exten => 1001,1,rpt(1001|P)

Assuming that 1001 is a private node. Not so ?

r(tone) or T (transfer) when by default the feature map has not been set.
Unless you did for a reason.

However, sloshing this around in the few brain cells I have left and looking at that dialplan stanza, I would say you would also need to have callerID name and number set because it is expecting it. You do that in the device sip stanza
That is a guess as I can’t see it. But it is looking for that info. I don’t use it connecting to my own nodes. I don’t want it showing up in my allmon.

Again, you would need to watch asterisk in the foreground to see just where it is going wrong or check the logs
/var/log/asterisk/messages

But there is no reason for my skinny version of the same will not function without CID info.
It’s not looking for it or using it.

Why not test just these 2 lines and see if it works and comment out the others for now.
exten => 1001,1,rpt(1001|P)
exten => 60694,1,rpt(60694|P)

After dialing, just command a *70 since there is no speech in the stanza.

But again, watch asterisk in the foreground for error.
asterisk -rvvv

10

I am sorry to have not read you correctly… LOL
I did not see the ‘OR’

The number you dial should be the node you are trying to connect to by number.
60694 in you case.

Seeing the1001 in the example/manual is some other phone extension.
You obviously do not need that except by example of some other phone extension.
If it were a node number, what I said would have been true.

Sorry to have confused you any. A Fault of bad eyes and speed reading I guess.

Looking at that manual, it states
The entry 1001 in extensions.conf is the extension number for your SIP phone. If you want to use a different extension number, change 1001 to your desired extension number.

11

Hi Mike,

I have tried all of you suggestions but I am still having an issue with it… I can connect to my node thru dialing ‘60694’ (my allstar node #) on my Cisco 525G2 phone but I get no audio out from my phone. When I dial ‘1001’ the extension I had set it for in ‘extensions.conf’ it just rings and doesn’t connect. I am trying to use extension 5 on my phone. I tried setting my extension on the ‘extensions.conf’ also to extension 5 and I still have the same issue.

When I type ‘asterisk -rvvv’ what type of error messages should I be looking for ?

maybe it is an issue with this particular phone model ?

Thanks,

-Tom
KE4QCM

12

You might also need to check the phone internal settings and make sure you have a codec in it set for the same as the extension setting. Preference to ulaw or g711u and gsm
I probably should have caught that in the beginning, but I failed to.

Well, it i a way to verify action are being taken in real time.
You should be able to recognize a error when you see it.

For instance when you dial on the phone you should see that attempt no matter what it’s dialing.

You need to look at if the phone itself is properly logged into the system.

So, from asterisk command line interface ( CLI> ), you should be able to type

sip show peers

And check the status of that phone. Is it logged in /registered ?
Would be a good starting point since we were off on a disoriented start.

The 1001 number is the extension described of the phone if you used the example from the manual. You don’t want to dial the phone you are dialing from. Perhaps it would ring if callwaiting is on. But would otherwise ring busy. But I’m not sure about the default settings for it.

13

Hi Mike,

I got my setup working now-the original setup instructions for allstarlink 3 with voip sip phone worked afterall and your troubleshooting steps helped also. Actually, I found out that if I moved my sip phone plugged into my main router directly the connection worked with a 192.x.x.x. address and now I have audio from allstar coming in/out ok to/from my sip phone. Originally I had my voip sip phone plugged into another router which gave it a 10.x.x.x address and that seemed to be causing the issue.

thanks again for all your help.

73,

Tom
KE4QCM

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