Asterisk PBX Twilio Elastic SIP Trunking Cisco hardphone

Dear Sirs,
I’m working on the setup SIP configuration with the assets of Asterisk server + Twilio SIP trunking + cisco hardphone + Twilio trial phone number.
Can you kindly provide a manual or video for the sip.conf connectivity?
Twilio provide pjsip manual but I already have sip peers and some knowledge about it.

Which Cisco desk-phones does you have exactly? I guess, that is your main issue, or at least going to be your main issue.

When it comes to new installations, you should start with Sangoma FreePBX. Or at least with chan_pjsip rather than chan_sip because those are actively and heavily maintained paths. If you go go for chan_pjsip, you mike be interested in Asterisk The Definitive Guide. Make sure you get the 5th Edition. If you want to stick to chan_sip, got for the 4th Edition.

Good time of the day,
I use CP 6941 with active load firmware SIP 9.4.1.3, the phone already has registered with Asterisk in the local network.
I think I have a mess on transport and a higher level of a global network:
Error communicating with your SIP communications infrastructure

Possible Causes

  • Your SIP endpoint is not reachable due to network connectivity issue between Twilio and your system
  • Your SIP endpoint is not responding (service down or maintenance)
  • There is a firewall in your network that is blocking SIP traffic from Twilio
  • Your SIP endpoint is sending an error response, such as SIP 500 response.
  • The SIP URI specified in your Trunking Origination URI, TwiML or REST API call is invalid

Thanks for the guide, of course, you are right about paths and recently manuals and forums discussion ability. But due to some point, I try to start my new Installation with SIP. I use a couple of YouTube and forums tutorials. For example one of the leader https://www.youtube.com/watch?v=jl9x9wlW4xk offer this kind of setup. And I use to learn connectivity. But I have difficulties with figure outing the value of the present lines. For example, in this tutorial, we have in sip.conf two trunks. But nowhere used credentials. How SIP endpoint can be registered and connect with the Asterisc server and how the Asterisk server can be connected with theTwilio trunk?

The service provider will normally provide typical configurations, which will normally work, but may not be the most secure or reliable. If your provider doesn’t provide Asterisk configurations, they are probably not in the right market for a naive use of Asterisk, e.g. they may be in the home user or SOHO market, rather than the medium sized office market.

Running more than one account on a provider is generally a bad idea; you should use a single account which allows enough simultaneous calls. If the provider only sells single call accounts (typically they say line, rather than call), they are almost certainly not in the PABX market.

The pjsip sample configuration file, which you should be using for normal new installations, has sections for typical SIP service providers.