Given that VoIP.ms is returning a misleading error response, please try these several changes, retest and paste a new log (with pjsip logger on).
Just finished applying all of the settings. Will be doing test calls and grabbing logs soon…
Here are the results for the change(s) in config and call attempts:
https://drive.google.com/drive/folders/1JzdUNa9BJQEkx7-UyzyiKZuimqkizZSu?usp=sharing
pjsip logger was off. Please turn it back on and paste another log.
No matter how you look at it, the call to 8004377950 (which is an ANI ‘verification number’ ) is not completing with a 'Cause 34". Given that, only voip.ms will be able to tell you why they won’t route it for you.
Do other numbers “complete” ?
Sorry about that. I’ll turn it back on and reattempt the calls in a few minutes.
Sadly, the same thing appears to happen for all outbound calls I make. I’ll try contacting VoIP.ms if the next logs I post don’t provide any new or useful info.
Have you paid your bills?
I’ve set it again:
I’m guessing that this method only sets it for until the next reboot. I’ll keep that in mind for future reference.
Probably not as effective as watching your calls with sngrep ,it’s much easier to seperate calls and decipher the SIP errors/failures on any SIP channel driver.
Yes. I also setup VoIP.ms to send a warning when the balance starts to get low.
I collected new logs, with pjsip logger enabled (same Google Drive folder from pervious post). New log files are named with a higher number at the end of their name (0, 1, etc.).
How would I use that command? Just curious…
Not so complicated, from a shell
sngrep
it’s a TUI so arrow keys pretty obvious “Method” indicated the nature of the transaction so in your case then look for “INVITE” , enter key will ‘drill down’ level by level , q and esc ‘drill up’
man sngrep for a manual
After bothering the support staff at VoIP.ms one more time, I received the following reply:
06:41 PM
Support Agent: Hello
How may I help you today?
Your calls are failing due to the Caller ID
Which has to be a 10 digit number
Specially to dial Toll Free numbers.
You are passing '338095_default'
06:43 PM
TopHatProdxns115: I think I've tried changing that before. How can I have FreePBX set the CallerID?
06:43 PM
Support Agent: This is a change you need to make on the PBX. Here is a guide you can refer to
https://wiki.voip.ms/article/PBXs#FreePBX_.2F_PBX_in_a_Flash
The phone number should be passed from your P-Asserted-Identity headers
06:45 PM
TopHatProdxns115: Will I be setting this in the Outbound Route, or the SIP Trunk settings?
06:45 PM
Support Agent: We do not provide advanced support for PBXs, apologies.
I do not have any further details
06:45 PM
TopHatProdxns115: All good. I'll look through the guide once more.
The guide that the support agent linked was for an older version of FreePBX, but I had already red through it at least twice before this interaction -
due to a previous agent linking it for a different suggestion, while troubleshooting this same issue
At this point, I ended up ignoring the guide and just going off of what today’s support agent mentioned.
After making those changes, I attempted an outbound call to a 10-digit phone number, with no + or country code. It went through. Now I just need to figure out how to get rid of the disgusting background noise XD
What noise are you talking about; your last post was the first use of the word ‘noise’ in this thread?
Does it occur on incoming calls? On all destinations? Before the call starts ringing or is answered?
From what I’ve encountered during testing, it seems to be present for both incoming and outbound calls where the call is answered by a human (not during automated prompt portions). I’ll have to check it again tomorrow, since it’s getting late. I’ll be looking into SMS at some point in the distant future, but the noise is the important thing (gotta be able to understand what people are saying during the call).
I managed to resolve the noise issue through the phone client.
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