I have setup SIP trunk. Asterisk info log showed that I am registered. However I am unable to receive inbound call despite I configured inbound route.
The same problem applied to outbound routes.
What do i do?
pastebin logs
Please help me with the directory. I am a newbie. Where do I get the log
The log shows no attempts at incoming calls.
Your log is garbled, possibly because you didn’t mark it up as pre-formatted text (</> on the formatting menu). However it appears that the call has a “DID” of 084363904, but your configuration doesn’t define that “DID”.
How do I define the DID?
I specified the DID (0843636904) in the inbound route configuration
FreePBX does not have a context named ‘from-sip’. I am guessing that your trunk configuration contains a line with context=from-sip (it should normally be from-trunk or left out), or you have specified an Incoming section (not normally needed) with USER Context set to from-sip (delete it or set it to from-trunk).
With SIP debug enabled, make another failing incoming call, paste the Asterisk log for the call at pastebin.freepbx.org and post the link here.
@Stewart1. thank you for your assistance.
after going through series of debugging I noticed the Incoming SIP setting was the issue.i cleared all the setting and I used only the Register String…
Also on the outgoing SIP setting, I included “fromuser” = user account, type=peer (instead of friend as stated by the provider)
Everything seems OK now.
Thank you so much.
If this is a new system, you should use chan_pjsip, as chan_sip is deprecated and scheduled for removal.
Unfortunately many providers copy bad configurations without considering what they actually mean, so, particularly, for chan_sip, you should assume that provider configurations are bad, although they usually work to some extent.
You are right.
currently, i discover, the trunk only allows one sip account and not the entire 30 lines we got.
I am caught in between creating 30 sip accounts or look for suggestions from the community members.
Any suggestion?
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