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Sangoma Phones 405 and 500 wont register
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Sangoma Phones 405 and 500 wont register
You posted a random log.
What did you specify in endpoint manager template for the SIP server?
https://wiki.freepbx.org/display/PHON/Auto-Provision+Quick+Start+Guide
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GV Cost Benefit
RAMEN… married once, on my 7th fionsai XD I never learn either
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no incoming call?! (noob alert)
Do a test with the Inbound Route DID Number blank. If that allows the call to come in, you either have the number in the wrong format, or the trunking provider is sending it in To or some other header, rather than in the URI.
For example, your provider may be sending 32345678, 4532345678 or +4532345678. The DID Number in your route must match exactly.
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no incoming call?! (noob alert)
Here is the problem, you have masked every vital piece of information that we would need to troubleshoot this. We have no idea if your Inbound Route is actually matching what is being sent because you’ve changed it all.
We have no real reference points. The DID being sent by the provider should be an exact match in Inbound Routes.
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no incoming call?! (noob alert)
I’ve only masked my phone number
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no incoming call?! (noob alert)
okkk if ANY set the phone rings 
so most there 
hm tried with my number with dk area code +45 no dice
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no incoming call?! (noob alert)
You mean the thing your provider is using to send calls and that has to match what is the Inbound Route? That thing? Yeah, we don’t know how they are sending it and how you have it actually set in the Inbound Route. Therefore we cannot truly what the issue is.
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no incoming call?! (noob alert)
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
> 0x6d500018 – Strict RTP learning after remote address set to: 91.217.201.94:18518
– Executing [8618610@from-trunk:1] Set(“SIP/86518610-in-0000000c”, “__FROM_DID=8618610”) in new stack
– Executing [8618610@from-trunk:2] NoOp(“SIP/86518610-in-0000000c”, “Received an unknown call with DID set to 8618610”) in new stack
– Executing [8618610@from-trunk:3] Goto(“SIP/86518610-in-0000000c”, “s,a2”) in new stack
– Goto (from-trunk,s,2)
– Executing [s@from-trunk:2] Answer(“SIP/86518610-in-0000000c”, “”) in new stack
> 0x6d500018 – Strict RTP switching to RTP target address 91.217.201.94:18518 as source
– Executing [s@from-trunk:3] Log(“SIP/86518610-in-0000000c”, “WARNING,Friendly Scanner from 91.217.201.94”) in new stack
[2018-11-13 19:43:23] WARNING[14734][C-0000000f]: Ext. s:3 @ from-trunk: Friendly Scanner from 91.217.201.94
– Executing [s@from-trunk:4] Wait(“SIP/86518610-in-0000000c”, “2”) in new stac
k
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no incoming call?! (noob alert)
Try setting up an “any/any” inbound route with no DID and no CID.
Send it to your extension/ring group.
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no incoming call?! (noob alert)
that works fine
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no incoming call?! (noob alert)
If an “any/any” route works, look at the logs and find a line that says something like “You should think about setting up an inbound route for this number” and set up an inbound route for that number. It needs to match exactly (country code, prefix, etc.) in order for the inbound route to pick it up from the inbound trunk.
Note that there is no shame in using an any/any route for all of your incoming calls. If that’s working and your other routes are not, having one can make the difference between something that worked for a while and suddenly doesn’t an inconvenience instead of an outage.
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no incoming call?! (noob alert)
guess I found the problem
Set(“SIP/86518610-in-0000000c”, “__FROM_DID=8618610”). <<---- thats the wrong number,
I must talk to my sip provider tomorrow, something is very wrong here,
Tests with another sip number (got 5) it worked right away, just by change I found it…
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Asterisk CLI output doesn't show call route
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no incoming call?! (noob alert)
This is why we don’t want people to mask this stuff. The very first line of this debug shows the exact problem (as you just pointed out). The wrong number is being sent.
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New install is totally unreliable!
Hi all, just installed FreePBX and haven’t even had time to set up extensions, all it has done is send me emails about updates every few days, but when I log into the interface, I have several issues:
- it is extremely slow - slow to load initially, slow to log in, slow to load the dashboard, slow to navigate. I’m talking 10-20 seconds per page. No it’s not an internet connection problem, it’s on the local lan.
- apply config red button is always showing - I always apply the config if I make a change (which there have been very few so far) so why is this always illuminated? And what changes is it actually applying?
- the dashboard isn’t loading the normal guff on the left side anymore
- updates to modules just crash it
please see attached images
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no incoming call?! (noob alert)
yar im writing my Sip provider now there is missing a 5 number,
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Asterisk CLI output doesn't show call route
Not sure what might cause this, you could try reinstalling asterisk with asterisk-version-switch
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Asterisk CLI output doesn't show call route
At the asterisk cli
core set verbose 3
it should be in /etc/asterisk/asterisk.conf under [options] as
verbose = 3
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New install is totally unreliable!
This is the distro? Firstly pm2 isn’t even installed. Secondly pages load fine for me on the same be setup. About 1 second.
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