I’ve worked a lot with Vega units - i suspect the problem is in your dialing plan.
If you ssh onto the vega, and type - log display on
then make an inbound call - you should see which “route” the vega is using - as a starting place.
If you then screenshot the inbound route - i suspect its a missing token
The word ‘Token’ lead me to finding ‘Token Help’ as a drop down on the dial plan pages. That in turn lead me to finding examples of settings that work to pass the caller ID to the SIP destination. For others reading in the future,
Source - enter "IF:02..,TELC:<.*>" #where 02.. is the source, like 0201
Destination - enter "IF:9901,TELC:<1>"
The observation here is that <.*> in the source collects the caller ID and <1> drops it on the destination.
Checking the vega log (by telnet, it didn’t like SSH, but the log was the same as I found through the web interface);
So, I now see the Vega is passing the Caller ID. Only problem is, FreePBX still shows ‘vega’ as the ID. I’ve spent a few more hours googling and reading and I know I’ll get there eventually, but if you can point me where to look I’d be grateful.
The best hint I’ve found in the FreePBX logs so far is;
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is 'vega' number is 'vega'
> dialparties.agi: USE_CONFIRMATION: 'FALSE'
> dialparties.agi: RINGGROUP_INDEX: ''
dialparties.agi: Methodology of ring is 'ringall'
Maybe ‘TELC’ is not the string FreePBC is expecting to parse as CallerID?
Name profile1
Interface ID 9901
Local Domain 10.18.6.231
Alternative Local Domain alt-reg-domain.com
From Header 'userinfo' Authentication Username
Contact Header 'userinfo' Calling Party
P Header 'userinfo' Calling Party
From Header 'host' Local Domain
To Header 'host' Local Domain
Redirection 'host' Local Domain
Transport UDP
Capability Set 2-voice+t38Udp
Reliable Provisional Responses off
DTMF Transport rfc2833
DTMF INFO mode1
RFC2833 payload (96-127) 101
On FreePBX
Trunk Name vega
Hide CallerID No
Outbound CallerID <our number>
CID Options Allow Any CID
Maximum Channels
Asterisk Trunk Dial Options [defaults = T, System]
Continue if Busy No
Disable Trunk No
PJSIP Settings General
Username Username is trunk name
Secret ••••••••
Authentication Inbound
Registration Recieve
SIP Server 10.18.6.230 (greyed out - left from when I tried Auth:outbound & Reg:Send/etc)
SIP Server Port 5060
Context from-pstn
I’m not sure of the reasons/difference between registered and unregistered, but I think it’s inferred that unregistered is less secure, so I’ve assumed it’s not connected with my lack of caller ID display.
I also looked at the inbound routes, saw the ‘CID name prefix field’ and put some text there. That text does get displayed on the extensions along with our trunk name (eg “TestCID:vega”). So at least I can be sure the extensions know how to display things.
Hi Steve
don’t know if you are still following this. I’ve been trying to do a similar set-up but not getting as far as you: FreePBX can see my Vega (endpoints shows it is available), but the Vega won’t register. In my log I get this:
2018-03-24 14:35:41] WARNING[8593] res_pjsip_registrar.c: AOR ‘vega’ has no configured max_contacts. Endpoint ‘vega’ unable to register
Are you able to look into your AOR file (pjsip.aor) and see if your vega has max_contacts configured? Because if not, then I need to find another reason. I have a feeling its something simple in my set-up of the trunk, but I have just followed all the wiki instructions, so don’t know what. Thanks.
How many contacts do you have set in your PJ-SIP config for the Vega? The error says “none”. Have you tried setting the number of contacts to a larger number (say 4)?
I tried, but not sure if I have done this correctly:
Using the built in file editor, I edited pjsip.aor_custom and put the whole of pjsip.aor inside it and edited this part:
[vega]
type=aor
qualify_frequency=60
contact=sip:192.168.0.110:5060
max_contacts=1
(ie added the last line to what I copied from the original unwritable pjsip.aor)
Should that have fixed it? It made no difference to the outcome. Got the same line stating :
AOR ‘vega’ has no configured max_contacts. Endpoint ‘vega’ unable to register
”res_pjsip_registrar.c: AOR ‘vega’ has no configured max_contacts. Endpoint ‘vega’ unable to register"
I always had that error. There was nowhere in the GUI to configure it that I found and everything worked, so I ignored it. I think I read it was a difference between pjsip and chansip but I can’t be sure of that
For outgoing calls, you decide by freepbx outbound routes, no?
I wrote myself lots of notes intending to share them but was never satisfied they were good enough. If it helps, this is the bit on Vega;
Connecting POTS/PSTN lines
Summary, connect the vega to the freepbx using registration mode: gateway, ID= name you’re calling trunk in freebpx (I chose ‘vega’) and password.
In Freepbx, add trunk, name it (I chose ‘vega’), set the ‘secret’ to the same as the password you made up for the Vega, Authentication: Inbound, Registration: Recieve. http://wiki.sangoma.com/Vega-Registering-Vega-PRIFXO-with-FreePBX
Note: The vega itself can be set to process calls in such a way as to route to different lines depending on the dial string.
Although, I have Authentication: Outbound, Registration:Send, SIP Server: The Vega IP Address. Port 5060
Apologies for formatting, copy pasting via my phone as I’m away fromy laptop and office
All sorted. The main thing I found was that I had to very carefully delete all existing settings (on the SIP page as well as in the Quick Setup page) before entering the new trunk name and password. The Vega did not do that well automatically (and I didn’t want to do a full factory reset).
So far I have only set up for incoming calls (BRI to Vega) to be routed to FreePBX using a simple dialplan in the Vega (altering the pre-set destination line in the “To_SIP” Dialplan to send the calls to the ip address of the FreePBX server. From memory it is something like IF:99…,TEL<1>,TA:192.168.x.x.
@sudoroo I hope things are still going well with your setup. I too am in the middle of a new PBXact (running FreePBX) with Vega 60G setup and 3 POTS coming into its FXO ports. With
From Header ‘userinfo’: Authentication Username
Calls pass through but show “vega” as the caller ID
When making this switch: From Header ‘userinfo’: Calling Party
Calls show in asterisk but do not get routed to the ring group. the follow asterisk message shows:
Request ‘INVITE’ from ‘"MY CID NAME " <sip:[email protected] >’ failed for ‘vega.ip:5060’ (callid: [email protected]) - No matching endpoint found
I haven’t made any other changes like the call plan. When I switch it back, everything works as normal except back to “vega” for the caller ID. Any thoughts?
If we assume you’ve set the system up the “standard” way (normal, out-of-the-box set up) you are mixing Chan-SIP (which is how you probably set this up) and PJ-SIP (which is where that “no matching endpoint found” message comes from). You probably need to set this up an PJ-SIP trunk for what you’re trying to do to work.
@cynjut this is setup with pjsip 5060 on both sides following the setup from the wiki. Calls run fine but don’t push caller ID; just the term “vega”. If I change the header on the vega device from username to calling party (in efforts to troubleshoot the problem), that’s when the calls fail hit the endpoint.
Nevermind @cynjut@sudoroo. I swapped the trunk to chan_sip and changed that header. Presto. I’ll leave the Vega trunk to PBX as chan_sip for now unless anyone has a reson to switch it back to pjsip.