List of SIP providers?

General Help
1

I have been battling with Broadvoice for a couple of weeks now. Intemittend service, my caller ID showing up as “Broadvoice”, and their “stellar” support for BYOD is making me rethink my decision to go with them. Is there a list or a comparison out there for different SIP providers?

I am looking to use my freePBX box with 2-4 incoming lines (I suppose that means 2-4 SIP trunks, right?) and I need to call Europe quite frequently, so I am looking for low cost outgoing calls to Europe. Any suggestions?

Thanks.

2

I have had good service with Flowroute. You can either buy a whole trunk, or just do per-minute billing without being limited to a specific number of trunks, which is nice so you’re not paying for trunks that you’re not using.

3

Here’s a list of providers that you might consider:

Sipstation - https://store.freepbx.com/store/
Callcentric.com
Voip.ms
Flowroute.com
Vitelity.com
Future Nine
Les.net

I’m not recommending any of these other than the first one, and I haven’t used all of them either, but these are the ones that I see mentioned over and over again in forums and elsewhere.

5

I always start with a restricted route for calls I don’t want to go out, using these dial patterns. It routes to a disabled trunk.

1900XXXXXXX
900XXXXXXX
1XXX976XXXX
XXX976XXXX
1976XXXX
976XXXX

Then I have an emergency route for 911 calls.

911
111

(I use 111 to test my emergency routes without having to dial 911). 111 is defined in every trunk to call a specific non-emergency test #, such as 800-555-8355

Then I have an interoffice route

+|*804X
+|*805X
+|4X
+|5X
+85|.

Then Speeddials

(2125551212) +222
etc.

The above allows you to dial 222 and your call will be delivered to 2125551212. Change the #s as you desire. I prefer speeddials here because it allows me to set a custom caller ID for them and set a specific route.

Then 411/511/611.

Then 800 calls

1800XXXXXXX
1866XXXXXXX
1877XXXXXXX
1888XXXXXXX
800XXXXXXX
866XXXXXXX
877XXXXXXX
888XXXXXXX

Then a route to dial 81 to dial my POTS Line 1

81|1NXXNXXXXXX
81|NXXNXXXXXX
81|NXXXXXX
81|X11

Then a similar route to dial my POTS line 2.

Then a similar route to allow 83 to force calls using one particular SIP provider, and another similar route for 84 for my backup provider.

Then a Caller ID Blocked route:

*67|011.
*67|1NXXNXXXXXX
*67|NXXNXXXXXX
*67|NXXXXXX
*67|X11

A force specific caller ID route (which has the Route CID override feature on to force a specific caller ID in a specific area code):

*212|011.
*212|1NXXXXXXXXX
*212|NXXXXXXXXX
1212+*212|NXXXXXX
*212|X11

(the second to last line allow me to dial *212 and then do 7-digit dialing).

And finally, a normal route:

1NXXNXXXXXX
NXXNXXXXXX
NXXXXXX
X11

7

If Broadvoice says that your trunk isn’t registered, then you probably don’t have your trunk settings done properly.

8

Registration is an issue that I have to fight with Broadvoice as well. The trunk works for days, then disconnects for no reason I can discern, and I have to go to a different proxy. Then it works again for a while. When I call them, they are very unhelpful, only tell me that the trunk is not registered. It might be stable now, but I have my doubts. If it continues this way, I have no choice but to switch as well.

I had one issue at the beginning where I could not get a registration. I had

insecure=very

in my peer configuration. I found that this has been deprecated and when I tried the settings below, I was able to register the trunk. My User Details are completely empty.

disallow=all
allow=ulaw
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=XXXXXXXXXX
secret=YYYYYYYYYY
username=XXXXXXXXXX
insecure=invite
context=from-trunk
authname=XXXXXXXXXX
dtmfmode=inband
dtmf=inband
nat=yes

Right now I am having a problem that I can call out, but some (only some) toll free numbers “cannot be connected as dialed”. BV is looking into that.

Good luck.

9

Several items about this config. the “user=phone” is not a valid keyword. Look at your peer in Asterisk, it’s generating an error.

You should also not need the insecure line, if you do the format is insecure=port,invite

username and authname seem like overkill.

Are you sure you don’t have a duplicate peer registering from another device?

It could also be a firewall/router issue with the NAT translation being torn down.

Here is the Asterisk docs and insecure:


;insecure=port                   ; Allow matching of peer by IP address without
 
1309                                  ; matching port number
 
1310 ;insecure=invite                 ; Do not require authentication of incoming INVITEs
 
1311 ;insecure=port,invite            ; (both)
10

Thanks for your hints. I took out the “user” line, the “insecure” line and “Authname” lines, reloaded, and now I am not connected. I’ will put them back in one by one and see what happens.

11

Ok, so I put the insecure line back in (as per your suggestion) and I am back on line. Still can’t call some toll free numbers, but that’s a different problem, I think.

Now, when I look at the Asterisk CLI and I enter “sip set debug on”, it seems to register every 20 seconds or so. I get this about

<— SIP read from UDP:206.15.156.221:5060 —>
SIP/2.0 200 OK
Call-ID: [email protected]
CSeq: 108 REGISTER
From: sip:[email protected];tag=as52eb4151
To: sip:[email protected]
Via: SIP/2.0/UDP 192.yyy.yyy.yyy:5060;branch=z9hG4bK023ffdf9;received=98.245.223.115;rport=5060
Contact: sip:[email protected]
Expires: 30
Content-Length: 0

I get this a few times, then it stays quiet for about 30 seconds, then it starts all over again. Is that normal?

12

Try voipvoip.com. I tried to use link2voip, but they were completely unresponsive when trying to sign up. Voipvoip does the per min charge with unlimited simultaneous calls. Works fine, just turn off g729 codec in your asterisk server. (They support it, but I think you have to license it from somewhere else).