NEW Tried SignalWire.com?

[billsimon]

I created an endpoint which has a username of:
[email protected].

then your username for the SIP trunk is 1234 and your domain for the SIP trunk is ABCD-randomstring.sip.signalwire.com.

So it looks like you just need to fix the username in the two places it appears.

Registration is Send

As you said you are only making outbound calls, you do not need to send registration. Set it to None.

I have not whitelisted sip.signalwire.com (although I can't figure out why I don't have to do this).

You don't have to whitelist anything because the provider is not initiating any connections with you, since you are not receiving calls from them. You are only sending calls to them, and your firewall permits you to send traffic outbound to anywhere.

 

[AndyInNYC]

I turned off media encryption and the call goes through!!


@billsimon,

So far, no joy.

My Project is 1234.
My Endpoint Name is [email protected]
My password at signalwire is !!!!###
My username at signalwire is ABCD


In PJSIP settings I have
Username as 1234
Secret is !!!###
Authentication set to Outbound
Registration is none

Advanced,
From Domain is the Endpoint Name
From User is 1234
Media Encryption is SRTP

Seems like everything is exactly as it should be, but and the PBX appears to place the call (cli shows the recording starting, etc.). My dialed cellphone doesn't ring.

Help?

Andrew
]

 

[billsimon]

I turned off media encryption and the call goes through!!

I doubt they are going to accept encrypted media signaled over UDP.

The only legitimate way to turn SRTP on is to also use TLS for SIP.

 

[mbrctv]

Hi, everyone,

I finally got signalwire working on my FreePBX with pjsip

I used their article as reference https://signalwire.com/blogs/product/connect-asterisk-signalwire-cloud-sip-connectivity.

I think the only things I changed was that I specified the AOR Contact and I use the sip server my endpoint

if you don't specify it sets it as: sip:<user>@<endpoint> but in their tutorial, they put it as: sip:<endpoint>

thank you all

 

[mbrctv]

I doubt they are going to accept encrypted media signaled over UDP.

The only legitimate way to turn SRTP on is to also use TLS for SIP.

from here it looks like they actually do

 

[billsimon]

from here it looks like they actually do

Pointless to turn on media encryption when you are going to share the keys in plain text allowing anyone to decrypt the streams.

 

[KNERD]

Anyone got a review of them yet?

 

[mbrctv]

Anyone got a review of them yet?

I used them for 1 month now, they were down twice for 30 min.. a few times calls where not routed to my PBX

I still like them, as i got unlimited channels

 

[KNERD]

thanks for that input. I guess they may be good for personal use for now, but with that many problems in just one month, it would not be good for business use.

 

[mbrctv]

thanks for that input. I guess they may be good for personal use for now, but with that many problems in just one month, it would not be good for business use.

someone on their slack sent me this:

I have had so many problems with SignalWire we are moving away from it after spending about 1 year with it, they also have a really bad inventory of DID numbers, it was great 1 year ago but not now.

 

[KNERD]

This is good to know. I just had to ask when the conversations about it all went dead. I would of thought it would of been better with the FreeSwitch folks involved

 

[wardmundy]

I've added a working SignalWire setup procedure in the new Incredible PBX Wiki.

p.s. SignalWire is no longer the "deal" it once was.

 

[Halea]

... they also have a really bad inventory of DID numbers, ...

This is a common issue with most development platforms including SignalWire, Twillio, Nexmo/Vonage, etc.
The issue is that their DID provisioning software can't track LNP. As users on those platforms port their favorite numbers to other providers those same numbers get automatically (actually when the user releases the number to avoid paying twice) recycled into the available DID pool. The new buyers of those numbers can make outgoing calls with spoofed caller ids but can never receive incoming calls because actually those numbers are routed elsewhere. They spend countless hours troubleshooting phone numbers which actually do not belong to them. The platforms eventually figure out and reimburse or substitute those numbers and manually remove them from their DID pools.
Some platforms are stubborn, they keep offering those same numbers well after a year they were notified by their own upstreams that they were removed from their inventory.

 

[Halea]

If anyone is interested in knowing what it takes to port out a phone number from SignalWire the attached image explains the procedure and the fee.