Sip.flowroute.com Not Registering

I am having difficulty getting sip.flowroute.com to register

Currently Free PBX 12.0.76 and Asterisk 11 are installed. Two phones are connected and can talk to each other
Thanks to Chris Sherwood with Crosstalk Solutions youtube video series on FreePBX 101 for getting me this far. ( I am brand new at this)

I configured the Trunk according to a Crosstalk video. I am using Flowroute for the SIP.

Below are the peer details code for the Trunk… Which came from Flowroute’s system configurator on their website

type=friend
secret=
username=
host=sip.flowroute.com
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
allow=ulaw&g729
insecure=port,invite
fromdomain=sip.flowroute.com

The register string is
3xxxxx7:[email protected]

The asterisk log reports about every minute that:
[2015-09-03 13:38:54] NOTICE[1840] chan_sip.c: – Registration for ‘[email protected]’ timed out, trying again (Attempt #302)

When dialing out there is a 20 second pause and then a message “The number is not answering”

I am able to connect to my server with Putty and have succesfully run a few commands. It looks like I need to learn Asterisk to know what to do here. Any tips for learning the commands would be helpful

I have read here that I might need to obtain a gl729 license.

Many thanks to any who can help. This forum is a great resource.

Hi!

Are you able to ping sip.flowroute.com, could it be a DNS resolution problem?

I use them and your configuration is slightly different from mine but I guess if it was made by their wizard it should be OK…

As for G729, yes you need a license and personally sound-wise I thought it was not worth it (it’s noticeably worse that G711)…

Have a nice day!

Nick

https://support.flowroute.com/customer/portal/articles/1849215-freepbx---prepend-your-tech-prefix-to-use-ip-based-authentication This maybe?

By the way, with Flowroute once the communication has been established the RTP streams do not come from known servers…

It is because of them I had to change quite a few things in my firewall configuration…

I use them for T.38 fax and they have been quite reliable…

I would try SipStation if it was available here (Canada) but it looks like they no longer have any DIDs here… All in all I currently use 4 different VoIP providers normally (not today though because I have a problem getting with Sangoma A200 card to work).

Good luck!

Nick

Canadian DID’s get snatched up quick. You can actually request one. More info at:
http://wiki.freepbx.org/display/ST/Special+Ordering+Local+DIDs+and+Toll-Free+Numbers

Good to know, thank you and have a nice week-end!

Nick

I am able to ping sip.flowroute.com

I ditched g729 for now and using ulaw

prepending my tech prefix did not work

I am getting the same behavior with a second trunk

Thanks to all who have helped. I am stuck at this point. Awaiting help from flowroute.

Cheers
S

OK… Flowroute says none of my packets are getting through But I am able to Ping sip.flowroute.com

There is another FreePBX server in the LAN using Port 5060… could there be a conflict?

FreePBX suggested changing to port 5061 for my second server… (Which I don’t know how to do right now). I could shutdown the other server possibly… What is the CLI ommand for shutting down gracefully?

Any suggestions for learning more is welcome. Most of the stuff out there is not helping me very much.

Regards,
S

You don’t need to shut the whole box down, on the “other box” issue

core unload chan_sip

to start it again

core load chan_sip

I have definitely the firewall that is causing the problem… All systems work fine when on a seperate LAN. I took the system home to my network and it worked fine

If you have a NAT’ed situation, then you can only forward 5060 to one server, so outbound connections from two servers on the same port will always be replied to at that address, generally that’s “not a good thing”

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Hi!

Do you have a subnet, could you give a separate IP to that server?

Have a nice day!

Nick