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.
Registration Retry Interval: 300 seconds
This is the information they provided to me to use on the FreePBX.
As Iāve never done this before Iād 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. Itās a UK number btw. Iām 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 Iāve 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 isnāt 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 Iād suggest using one of the established, UK, VoIP providers. If you are locked in, Iād suggest modifying your subject to include the brand name, as you really need input from a successful user of this service.
I couldnāt 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.
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.
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 āPJSIP syntax error exception when parsing āRequest Lineā on line 1 col 1ā. When I googled it the only thing I found was for Cisco 7941 and thatās 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 canāt 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 Iāve deleted the trunk and added it again. When I put for āNameā 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 āsip:voip.sogeavoice.com:5060ā on registration attempt to āsip:[email protected]:5060ā, retrying in ā60ā.
Then I changed āSogeaā 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 hasnāt 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 donāt 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 āRequest 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<transportLayerProtocolā and change it from 2 to 1 and then restart the Cisco phone. (it needs to have ā>ā, but if I leave it here itās not showing the whole text)
If they havenāt 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 donāt provide access to the account details,so you canāt 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).
@david55 my provider is No One Internet/Home Telecom going through a dark fibre in OpenReach network in my area (donāt ask, everything they made complicated ). 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 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 canāt 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 āpjsip 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.
Based on everything Iāve seen posted and the fact that @Stewart1ās original help post didnāt 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, thereās no associated outbound_auth config to pull the credentials from.
@darkness please make sure the trunk settings have Authentication set to Outbound.
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.
Hi @Stewart1, @BlazeStudios, @david55
Please find the attached log with the ālogger 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 donāt 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.
). 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