Flowroute New POPs Chan_Sip

FreePBX Providers
1

I switched a couple servers over from the sip.flowroute.com to us-west-or.sip.flowroute.com still using Chan_Sip Trunks, everything appears to be working fine for a couple months now. I get the Flowroute email today saying PJSIP trunks are needed for the new servers, to deliver the calls over multiple addresses.

I am registered to the 34.210.91.112 server, I just assumed the calls would only be delivered from this address and I would only be missing out on the failover benefits of the POPs. I havent had any problems with incoming calls that I am aware of, just curious if anyone else is still using a Chan_Sip trunk with the new POPs?

2

I would suggest going to PJSIP for this for a couple reasons.

  1. A PoP (Point of Presence) does not mean one server. Each PoP has about 26 IPs related to it and you can receive calls from the PoP you are registered with. That means any of those 26 IPs.

  2. Chan_SIP is horrid at SRV. It only uses the first record, so if that fails it will no look at the next one.

Plus Chan_SIP would require 26 peers to make sure all the IPs are resolved properly. PJSIP would take one endpoint because it does SRV properly and can get all the IPs of the SRV record to match against.

3

Thanks for the input @BlazeStudios, I was just kind of curious also because Flowroute’s Support reply was “I am either just Extremely Lucky its working or have allow anonymous sources on”, which I dont.

But I do plan to convert the trunks.

4

They’re right. You’ve been extremely lucky.

5

@BlazeStudios one last question, I seen you have to register the PJSIP trunk with 5060. But I can still have a non-standard bind port under Asterisk Sip Settings?

Thats how I read this from them…
“You can use any source port on your end you wish, but you must send to the destination port 5060/5160 for udp or 5061 for tcp/tls.”

Thanks!

6

Right, you can listen on any port you want. They listen on 5060. So that is where your requests need to be sent to. So in a PJSIP trunk the SIP Server would be us-west-or.sip.flowroute.com and the SIP Port to 5060.

7

Thanks Tom I appreciate your help!

8

Updated both servers, everything appears to be working. Except the Asterisk statistics hour graph is blank all the sudden on both.

Freepbx 14(Ill check the exact version tomorrow)
Servers are probably 2 months out of date on updates. Ill update tomorrow see if problem persists.

9

Note that you need to white list whatever IPs/Networks in firewalll for whatever entries you put in the PJSIP trunk “Match” field.

10

Thanks @lgaetz, everything appears to be in working order, I came in this morning and the statistics was back working also.

11

Yeah, so this is going to be addressed right?

12

https://issues.freepbx.org/browse/FREEPBX-18741

13

Please don’t take this the wrong way but linking a ticket that was opened almost 14 months ago, got updated with nothing real 12 months ago and still is sitting untouched and unassigned in the queue a year later does not show me it as being addressed it shows the opposite.

All I see now is a ticket that has been in the flow for 14 months, not one developer taking ownership and none being assigned ownership.

14

@BlazeStudios I didnt see any documentation on Flowroute for the TrustRPID and SendRPID, I typically used those on Chan_SIP trunks, is the still common practice on PJSIP trunk too? If so, would “Both” on the Send be appropriate.

Many thanks for your input

15

For PJSIP set the Send set to “Both” means RPID and PAI will be sent as headers.

1 Like
16

Thank you Sir

17

This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.