Takes 3 or 4 calls to connect to the pbx system - voip.ms

this is happening on multiple systems, been a while…

FreePBX 16.0.40.8
Asterisk Version 18.20.2

Provider: voip.ms

just ported abunch of our numbers to voip ms, and have them forwarded to our Freepbx system – 1 live system - 1 fresh install system for replacement…

they are both doing the same thing… it takes at times 3 or 4 tries to get connected, it will return a “customer is not availible”… message or sometimes 3 quick busy signals and then hangs up. I thought it was only our live sysetm, but on my new replacement system the same issue occurs.

the trunk on the live system is actually connected to an Allstream Trunk providing our main phone number

in this time, there are no “hits” to the pbx, the only activity happens when it finally connects.
then it will work,t hen it will go back to its “customer is not availible” routine"

ths is happening with both SIP and PJSIP trunks…

spoke to voip.ms and asked for any recommended settiings, he noted:

“I’d suggest setting “Registration Expires” to 3 or 5 minutes on your device/system
and having NAT Keep Alive enabled”

are these flags availible in the trunk “User Details”. I’m using a SIP trunk.

has anyone experienced this and have any possible settings that may help eleviate?

thanks,
Peter

here are the sip settings:

username=XXXXXXXXXXXXXX
type=peer
trustrpid=yes
sendrpid=yes
secret=XXXXXXXXXXXXXXX
qualify=yes
nat=yes
insecure=invite
host=toronto9.voip.ms
fromuser=XXXXXXXXXXXXXXXX
disallow=all
context=from-trunk
canreinvite=nonat
allow=ulaw

“…he noted:”

“I’d suggest setting “Registration Expires” to 3 or 5 minutes on your device/system

and having NAT Keep Alive enabled”

If voip.ms use registration id you should change type parameter.

Configuring SIP - Asterisk: The Future of Telephony [Book] (oreilly.com)

thank you, I will ask them. appreciate the document.

Are you actually registering to voip.ms?

On the VOIP.ms web page, it seems it use registration Id and service is provided through any internet connection and it need to pointed to a server by choosing server closest to account localization.

https://wiki.voip.ms/article/Getting_Started

yes, the truck is “Registered” successfully – no problems with that…

its when you call the #'s you get the unavailable message… try 3-4 times then freePBX kicks in… in those “unavailable” calls – there is no activity or attempted connections to the pbx box.

I’ve decreased ring count, increased it, tried every option availible… and its happening on multiple pbx’s and alsoour live pbx is getting its trunk from Allstream, and it also experiences this behaviour…

thanks,
Peter

yes, that is all setup and functioning correction, the trunk is created and register with the asterisk server…

Are these systems behind a firewall/nat?

Hi Tom,

yes, both are behind traditional FW’s, no aws or azure hosting, the live is in a datacenter and the other to replace it is in my “test” environment.
both use different internet providers, and aer vm’s running on VMware ESXi

my FW is a Sonicwall TZ300, the data center I don;t know as the MSP manages this and I don;t have access to this setup…

thanks for your time,
Peter

Disable source port remap and set the UDP timeout to 300 (5 minutes).
You may find this useful: https://www.reddit.com/r/freepbx/comments/qun72m/freepbx_with_sonicwall_firewall/

If you still have trouble, capture traffic on the WAN interface of the firewall. On a failed call, do you see the incoming INVITE? If so, and it doesn’t show up at the PBX, the firewall blocked it and you should use its tools to find out why. If the INVITE doesn’t appear on the WAN, look at the outgoing REGISTER requests to see whether the Contact header has the correct public IP and port.

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If the trunk is in peer mode, is no sense user and password, since a registration request will not sent.

A chan_sip type of peer means that it makes both inbound and outbound calls. A peer won’t accept an inbound call unless the IP matches the host. If a register string exists in the config, then a registration is made.

Hi Stewart,

I have implemented your recommendations and am doing some testing… it appears this has possiblity resolved another issue I was having with the no audio on the sangoma Desktop App. made teh changes as per the document, and lone and behold Sangoma Desktop began to work on an “external” network

just testing the “takes 3 or 4 calls”, this appears to have stabalized… but then it could just be good run… as for any any INVITE or RGISTER messages hitting the PBX… nothing comes through in this “not available” window, nothing at all. then after the 3rf/4th/5th attempt it would just pick up as if nothing ever happened.

voip.ms wasn’t very helpful in their troubleshooting with this, surprised they don;t have many more customers calling about this.

This isn’t a Voip.ms issue, it is a ‘your firewall’ issue.

Hi Tom,

I believe you are correct, the system is now picking up normally after the changes made from this thread.

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