Setting up new trunk - Help required

Hello boys and girls,
I have a FreePBX 17/Asterisk 21 running on a SuperMicro server and FreePBX 16 / Asterisk 20 running on Raspberry Pi 4. Both of them work pretty good on internal calls with 5xCisco 7975, 1xCisco 7941 and 1xYealink T42S. Until now I have never used the whole system for inbound and outbound calls, but it was in mind. My ISP offered me a free VoIP, so I accepted.

  • Primary Registrar: voip.sogeavoice.com
  • Primary Registrar Port: 5060
  • Username: 077*033
  • Password: different
  • Primary Proxy: voip.sogeavoice.com
  • Primary Proxy Port: 5060
  • Expire Time: 300 seconds
  • Registration Retry Interval: 300 seconds
    This is the information they provided to me to use on the FreePBX.
    As Iā€™ve never done this before Iā€™d really appreciate a step-by-step guide how to set up this trunk for use. As the username is my phone number I changed few numbers with * to avoid spam calls from the bots crawling around and also changed the password. Itā€™s a UK number btw. Iā€™m still crawling in the SoGEAVoice web page to try to find a login page for this username and password and try to see if there are any logs there or anything, but still no luck.
    I will be very thankful to anybody who can help me with a step-by-step guide how to config this as Iā€™ve never done it before.

This information is incomplete, so one would need to try various guesses. Whilst one might guess right first time, it may take several attempts, and also examining incoming traffic, to work out the full details. (For a start, if the proxy is really needed, you need the domain name for their actual service endpoint, not just the proxy. I suspect there isnā€™t a proxy, or it is invisible to you. There are also questions about how you identify yourself, what format caller ID is in, do they send DID, or just echo your registration address, etc.)

Are you locked into this provider in some way? If not Iā€™d suggest using one of the established, UK, VoIP providers. If you are locked in, Iā€™d suggest modifying your subject to include the brand name, as you really need input from a successful user of this service.

I couldnā€™t find any configuration information on their web site, only FUD material about digital switchover. I also have concerns that they have chosen the brand name for poor reasons.

This seems to be straightforward ā€“ just give it a try. Set up a pjsip trunk with these settings:

Trunk Name: sogeavoice
Outbound CallerID: (your Username)
Username: (your Username)
Secret: (your Password)
SIP Server: voip.sogeavoice.com
General Retry Interval: 300
Expiration: 300
Contact User: (your Username)
From Domain: voip.sogeavoice.com
From User: (your Username)

Leave other settings at their defaults.
Set up your Outbound Route and Inbound Route as desired.
If you have trouble:
At the Asterisk command prompt, type
pjsip set logger on
If the trunk is not registered, paste the Asterisk log for a registration attempt (including any replies) at pastebin.com and post the link here. If registration is ok, paste the log for a failing attempted outbound and/or inbound call.

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[2024-11-15 17:02:11] ERROR[3801041] res_pjsip.c: Unable to create outbound OPTIONS request to endpoint SogeAvoice
[2024-11-15 17:02:11] ERROR[3801041] res_pjsip/pjsip_options.c: Unable to create request to qualify contact sip:7733*@voip.sogeavoice.com:5060 on AOR SogeAvoice

This is what I get in the System Logs. Can you please let me know from where to enable the logger on asterisk?

login to the freepbx server as root, type asterisk -rvvv
At the CLI> prompt, type Stewarts above commands.

You may also need to : pjsip set debug on
( to turn those commands off, run the command again but replace word on with off )

After enabling the logger on asterisk, this is everything I got in logs:

Please help!

At the Asterisk command prompt, type
pjsip show aor SogeAvoice
and post the output.

@Stewart1 hereā€™s what came out

There seems to be a strange entry for Outbound Proxy. In the trunk settings, this would normally be left blank. Do you have something there? If so, why?

@Stewart1 this has been given to me by the ISP and as I never did something like this before I entered all the data.
Funny thing is that after I entered all that data for the trunk I started getting errors ā€œPJSIP syntax error exception when parsing ā€˜Request Lineā€™ on line 1 col 1ā€. When I googled it the only thing I found was for Cisco 7941 and thatā€™s too old and not supported any more blah blah blah. The really funny part is that I have 1xCisco 7941, 5xCisco 7975 and 1xYealink T29G and the Cisco 7941 and the Yealink T29G are working, just the newer 7975 canā€™t register and making issues, where the really old one is connected, registered and working without issues. Any idea if this could be connected to the Trunk and how to fix it?

@Stewart1 Iā€™ve deleted the trunk and added it again. When I put for ā€˜Nameā€™ to be Sogea it started to try to connect as [2024-12-06 16:56:41] WARNING[3511861] res_pjsip_outbound_registration.c: No response received from ā€˜sip:voip.sogeavoice.com:5060ā€™ on registration attempt to ā€˜sip:[email protected]:5060ā€™, retrying in ā€˜60ā€™.
Then I changed ā€˜Sogeaā€™ to my actual username, but unfortunately the outcome is the same. Does this mean that the trunk is not active at the other end or I did something wrong again?

Typically it means that voip.sogeavoice.com:5060 is not a valid address, or there is a router configuration error, or your ISP blocks the use of SIP. It either hasnā€™t got as far as worrying about the validity of the account, or they have implemented a protocol violation in the way they implement rejecting an invalid account.

I still donā€™t understand why you are not using one of the well established VoIP providers.

@david55 this has been given to me for free from my ISP as part of my Gbps fibre connection and as per their words it should be set up and working.

"PJSIP syntax error exception when parsing ā€˜Request Lineā€™ on line 1 col 1ā€. - I managed to fix this one.
For everybody suffering with this error on Cisco phones, go to your config file (SEPMAC.cnf.xml), find ā€œ<transportLayerProtocolā€ and change it from 2 to 1 and then restart the Cisco phone. (it needs to have ā€œ>ā€, but if I leave it here itā€™s not showing the whole text)

If they havenā€™t provided more detail than that, it sounds like they have provided a phone as well as the service, and only support the combination. This is what BT do for domestic subscribers,.and they donā€™t provide access to the account details,so you canā€™t substitute your own phone.

Incidentally using SoGEA in the name is misleading, as SoGEA is FTTC, so would not be presented as fibre.

You can use pre-formatted text markup to avoid this problem (</> button).

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@david55 my provider is No One Internet/Home Telecom going through a dark fibre in OpenReach network in my area (donā€™t ask, everything they made complicated :smiley: ). As I was told by their IT department they just bought the VoIP service from SoGEA, linked it to the service they provide me and ā€¦ Voila! Yes, but No :smiley: Please take a look in the Asterisk LogConnected to Asterisk 21.5.0 currently running on voip (pid = 2702 - Pastebin.com log. I included few things there and my dialplan.xml (copy-paste from internet). Maybe I did something wrong and I canā€™t catch it, but you might.

[2024-12-09 13:43:40] ERROR[4074416]: res_pjsip_outbound_authenticator_digest.c:450 digest_create_request_with_auth: Host: '93.95.124.116:5060': There were no auth ids available

The log is incomplete, and you need ā€œpjsip set logger onā€ to be in effect, for full details, but this is saying that they want you to prove you know the password, but you have no passwords set for their service.

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Based on everything Iā€™ve seen posted and the fact that @Stewart1ā€™s original help post didnā€™t include the auth settingsā€¦it seems the trunk is created without authentication set. Reviewing the current trunk settings, outbound_auth setting is missing.

So when this trunk is being challenged for auth, like @david55 said, thereā€™s no associated outbound_auth config to pull the credentials from.

@darkness please make sure the trunk settings have Authentication set to Outbound.

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The settings I suggested in post 3 of this thread were based on everything not mentioned to be left at the defaults. Registration would be left at Send, Authentication would be left at Outbound, Outbound Proxy would be left blank, etc. Did you try this? If so, what was the result?

I recommend that you delete the trunk, set it up again as I suggested, turn on pjsip logger, make a test call from a device known to not cause trouble (such as your Yealink), paste the Asterisk log for the call and post the link. With luck, it should be easy to see what is missing.

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Hi @Stewart1, @BlazeStudios, @david55
Please find the attached log with the ā€œlogger onā€.
@Stewart1 I deleted and re-added the Trunk again as per your post and as per @BlazeStudios suggestion. The log has been taken after that and the call was made from the Yealink device (the Cisco devices are working fine as well). Obviously I had to hide my IP, my number and dialed number with ***, as I donā€™t want all those lurking bots to get them.
@BlazeStudios I do have a password for the trunk and it has been set in the config. I can provide screenshots of the config as well if needed.

Thank you very much for your time, patience and help.

The number sent from the Yealink, which you represented as 8 asterisks, did not match any Outbound Route pattern.

For testing, you could set up a route with prepend and prefix left blank, match pattern
X.
and CallerID left blank.

Make a test call to a number you donā€™t have to redact, such as a local McDonaldā€™s, paste a new log.

Dial the number in the same format that the trunk expects.

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