PBX, Asterisk and CTI

Hello All,

I work for a company in NL called UPTIC and this is not a sales thread but for any one out there who is trying to setup Asterisk, CTI and Phone support for zammad feel free to contact me on this thread, my hopes is that we can turn this into a full open guide.

Here are some of the building blocks we deployed for getting asterisk integrated with Zammad:

PBX: we use asterisk 16 with WebRTC and Asterisk Manager Interface for calling
in asterisk we use queues for incoming support calls, we also use SRTP and TLS for encryption of calls.

WebPhone: we use an adapted version of ctxSip (a web rtc phone client) that you can find, use and download from here opensource!: https://git.uptic.nl/uptic-public-projects/uptic-web-phone

Mobile VOIP we use Zoipher (because we need SRTP encryption for calls)

Desk Phones We use Yealink T41S (also using SRTP encryption)

CTI Bridge: We use this amazing CTI Bridge from Martin von Wittich: https://github.com/martinvonwittich/asterisk-zammad-cti-bridge

For trunks we use Twilio but any sip trunk will work!

For all of you out there who want to get this running or something similar but are a bit stuck on PBX config or want to use web phones lets try to make this a starting point for documentation so everyone can use it!

Maybe it also would be a good idea to migrate this to the documentation if it’s a general thing for asterisk.
I think that would help users to really benefit from it. :slight_smile:

I think that would be really awesome :slight_smile:

Can anyone edit the docs as I could always make a start to the base setup and adapt on feedback.

At one point would be awesome to integrate the webrtc phone into the portal but I need to look at how to do that:)

This topic was automatically closed 120 days after the last reply. New replies are no longer allowed.