NO JOY Problem configuring IPBX13 to use flowroute's new pops.

piaftnt

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Has anyone successfully configured IPBX13-13 to use Flowroute's new pops. They have an article at
https://developer.flowroute.com/docs/inbound-and-outbound-calling-with-flowroute-new-pops/#utilizing-new-pops-for-outbound-calls but I do not understand how that relates to freepbx. I have tried adding the tech prefix as described at https://support.flowroute.com/SIP_Trunking_and_Voice/PBX__Configuration_Guides/FreePBX/Prepend_your_Tech_Prefix_to_use_IP-Based_Authentication_in_FreePBX but still no joy.
 
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I just did this today on Incredible PBX 13.
Some gotchas I encountered:
Needed IPTABLES entries for new pops.
Had to create new Flowroute trunk (not use existing) due to my entensions being PJSIP. You must create a PJSIP trunk if extensions are PJSIP
Had to change the ports used in SIP settings. Default is SIP @ 5060 and PJSIP @ 5061. I needed PJSIP on 5060. Do this in Asterisk SIP Settings.
Use registration for routing at Flowroute. IP routing would not work.
When configuring registration in PBX be sure to only select Outbound. If Inbound or Both is selected it messes with the trunk login name.

Hope that helps.
 
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Mostly because it is newer technology but that really is not a good reason. There are new featrures and programming reasons for PJSIP. I was trying it first for multiple phones on a single extension to replicate old analog shared lines in small businesses that cannot seem to grasp or don;t like Park features. I don't remember finishing that testing but PJSIP in use ever since. This link explains it better:
https://www.linkedin.com/pulse/faqs-sip-vs-chansip-chanpjsip-kent-adams

Excerpt from that link:
What is CHAN_SIP?


Chan_sip is a channel driver used for SIP functionality in Asterisk based devices (and likely others) for years. A channel driver is what allows your device/software to communicate via some protocol (SIP, IAX, Skinny, etc). Chan_sip was developed when SIP was fairly new and prior to 2014 if you were communicating via SIP it is extremely likely that some device in that conversation was operating with chan_sip.

It is the only SIP channel driver in Asterisk version 11 and lower. Starting in Asterisk version 12, you have access to chan_sip and chan_pjsip. Many people are still using chan_sip because it is well known, stable, time-tested, and supports all of the features they need for regular SIP communications. However, it is not easy to modify for new feature support, and is going to be surpassed by chan_pjsip in the future, in terms of number of devices using it, and number of developers extending it, customizing it, and reviewing it.

What is CHAN_PJSIP?

PJSIP is a library which has become the foundation for the chan_pjsip channel driver in Asterisk version 12 and higher. You can use chan_pjsip by itself, or in parallel with chan_sip (if you know what you are doing). PJSIP is very modular and a change to one module does not affect the others. In other words, the developers of Asterisk (or any other communications platform using PJSIP) can add (or remove) features much more easily and have less risk of introducing a bug.

Presently chan_pjsip is relatively new, yet an increasing number of Asterisk based systems are choosing chan_pjsip as the default SIP channel driver. Ultimately it doesn't require much adaptation from end-users on GUI based systems (FreePBX, Yeastar, Grandsteram, etc), but it does have a small learning curve for non GUI systems users (see this website for an overview and some examples).

The current feature set for the PJSIP library can be found here

If you want to get more detailed information or even inspect the source code yourself you can visit www.pjsip.org

Parting Thoughts:


If you need some new SIP feature, or are using some newer module on your Asterisk server, you are likely going to choose chan_pjsip out of necessity. If you'd like to see what all the buzz is about, but are concerned for your safety; just remember, you cannot break SIP, and chan_pjsip should work for pretty much every use case where chan_sip already works. If you are just using SIP for VoIP and don't need anything beyond typical telephony features chan_sip should work for you for years to come.
 

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