DEAL Tried SignalWire.com?

w1ve

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From the makers of Freeswitch. Features similar to Twillio...

But:

US DIDs are $0.08/month
Termination is $0.00255/min
Origination is $0.09/min

With full automation API.

Anybody playing with it?
 

AndyInNYC

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I think Origination is off by a factor of 10 - I think it's $0.009/min. Cheap enough to be called free (almost).

Andrew
 

wardmundy

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I think Origination is off by a factor of 10 - I think it's $0.009/min. Cheap enough to be called free (almost).

Andrew
Didn't much matter because it wouldn't connect to an Asterisk server for incoming calls until about an hour ago. I'll watch the call log for actual pricing.
 

wardmundy

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@AndyInNYC: Looks like you get charged for 2 legs on incoming calls: one to the SignalWire server and a second cheaper SIP relay to your Asterisk server or softphone that registered to SignalWire. Same is true in reverse with outbound calls.

Here's a 1 minute, 1 second incoming call from the log using my cellphone to place the inbound call:

 

krzykat

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What false advertising. Is that how twilio bills also? So essentially - they're charging you for the carrier cost at .00255 (as advertised) - but then they're charging you a connect fee of .0007 - making the real cost .00325
 

w1ve

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Yup. False Advertising. Here's a 1-minute 2-second softphone call to a DID on another Provider connected to my Asterisk box.
They don't talk about the "Relay SIP Call" in the Advertising.

No vendor uses a model like this. So they are trying to cover the integration services cost with this.

Oh well.

Still early in the game, but I bet they will not change this.

signalwire.png asterisk.png
 

krzykat

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Yes - and forgot the other observation that billing is not 6/6 but 60/60 - that adds up too.
 

wardmundy

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And just to dot all the i's, here's what happens when you dial from one extension (endpoint) to another:



Also looks like you get double-ding'd for failed/unanswered outbound PSTN calls as well...



Only thing that is "free" seems to be the SIP registration transaction(s).
 
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wardmundy

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New pricing with full minute rounding. Prices shown do not include the $0.0007/min. additional charge for the extension/endpoint connection between your softphone or PBX and SignalWire on all incoming and outgoing calls. Pricing applies to all calls including failed/unanswered calls.

 

wardmundy

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DIDs aren't bad either. I know it's not their desired use case, but a big family spread out all over the place would be a bargain on this platform with SIP phones. Each could have an extension with a dedicated DID for $0.08 a month. The calls to each other would be dirt cheap, and regular calls and SMS messaging are dirt cheap, too.

Outbound calls: $0.0072/min.
Incoming calls: $0.00325/min.
Ext-to-Ext calls: $0.0014/min.
 
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tycho

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Didn't much matter because it wouldn't connect to an Asterisk server for incoming calls until about an hour ago.
Ward, could you post the (sanitized) settings that you used to get inbound working on Asterisk? I haven't yet cracked that code. And in trying to do so I may have broken my sandbox so I've got to run out and spin up a new one.

(brg on DSLR)
 

AndyInNYC

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Wow.

I'm paying $0.012/min with flowroute, and given their service I've always thought of them as 'cheap enough'.

If I had any volume at all I don't know how I'd justify staying with them.
 

wardmundy

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Ward, could you post the (sanitized) settings that you used to get inbound working on Asterisk? I haven't yet cracked that code. And in trying to do so I may have broken my sandbox so I've got to run out and spin up a new one.

(brg on DSLR)
On the SignalWire side, be sure to turn off top 2 ciphers for your endpoint and enable only G722 and PCMU codecs.

Using latest Incredible PBX 13-13 for CentOS (without GVSIP code), here are the PEER Details we're using with a standard registration string:
Code:
type=friend
trustrpid=yes
sendrpid=yes
secret=your-endpoint-password
qualify=no
insecure=port,invite
host=yourspace-xxxxxxxxxx.sip.signalwire.com
fromuser=your-endpoint-name
fromdomain=yourspace-xxxxxxxxxx.sip.signalwire.com
disallow=all
defaultuser=your-endpoint-name
context=from-pstn-e164-us
canreinvite=no
allow=g722&ulaw

Registration String: your-endpoint-name:[email protected]/10-digit-DID
2/23 UPDATE: $55 signup credit at SignalWire with promo code: ITEXPO2019
 
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wardmundy

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Wow.

I'm paying $0.012/min with flowroute, and given their service I've always thought of them as 'cheap enough'.

If I had any volume at all I don't know how I'd justify staying with them.
AS A CAUTIONARY NOTE, TREAT SIGNALWIRE AS A SANDBOX ONLY PROVIDER & EXPERIMENTAL PLATFORM AT THIS POINT.
 

tycho

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On the SignalWire side, be sure to turn off top 2 ciphers for your endpoint and enable only G722 and PCMU codecs.

Using latest Incredible PBX 13-13 for CentOS (without GVSIP code), here are the PEER Details we're using with a standard registration string:
Thank you! I just created a new CentOS PBX 13-13 instance so I'll work on getting things running there.

** EDITED TO ADD **

In Signalwire do I also set "* ENCRYPTION" to "optional" or "default" or "required"?
 
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PBXinmyhouse

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I chose OPTIONAL.
I have within the last week, added a sandbox Asterisk 13 (with GV and all the bells and whistles) CentOs and am having a heck of a time getting SignalWire to work.

I have already gotten signal incoming/outgoing working on a Obi202 device, and i was hoping to use that to get around the encryption requirement having it act as the middle man for my old asterisk (still production) but, that has failed.

With Asterisk13, i cannot get outbound or inbound to work.
I changed the qualify=no to qualify=yes to validate and make sure i was registering, as i see no way of getting any log information on signal wires side. PBX13 shows it is successfully registered.

Here is the log on an outbound call (i edited out the dialed number)

] pbx.c: Executing [[email protected]:23] Dial("SIP/1001-00000007", "SIP/SignalWire/1******5988,300,T") in new stack
[2018-12-26 10:33:47] VERBOSE[8079][C-0000000c] netsock2.c: Using SIP RTP TOS bits 184
[2018-12-26 10:33:47] VERBOSE[8079][C-0000000c] netsock2.c: Using SIP RTP CoS mark 5
[2018-12-26 10:33:47] VERBOSE[8079][C-0000000c] app_dial.c: Called SIP/SignalWire/1******5988
[2018-12-26 10:33:54] VERBOSE[8079][C-0000000c] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
[2018-12-26 10:33:54] VERBOSE[8079][C-0000000c] pbx.c: Executing [[email protected]:24] NoOp("SIP/1001-00000007", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 58") in new stack

Any thoughts?
Ps, running - Incredible PBX 13.0.192.19 'Incredible PBX Server'
Current Asterisk Version: 13.18.4

I want to also add im getting charged on SignalWire for 'Relay Call' even though i just get a busy signal.
 
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tycho

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After endless tinkering I have finally been able to receive incoming calls to my Signalwire DID registered to my IncrediblePBX CentOS 6.9, "13.13" instance.

A couple of quirks:

I have Enable TLS = yes and SSL Method = sslv2. That's correct, or necessary?

I can seemingly ONLY pull this off if I set Allow Anonymous Inbound SIP Calls = yes and Allow SIP Guests = yes. I don't ever do that. Is there a way around this that I might be missing? Otherwise, the PBX complains "WARNING,"Rejecting unknown SIP connection from 104.248.176.184"" even though I have added the following Signalwire-related IP addresses (ones that I am aware of) to IPTables (added to the "trusted providers" in /usr/local/sbin/iptables-custom):

/usr/sbin/iptables -A INPUT -s 104.248.176.184 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 159.65.244.171 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 178.128.235.81 -p udp -m udp --dport 5060:5069 -j ACCEPT


The PBX will also complain "Declining non-primary audio stream: audio 19022 RTP/AVP 0 101 13" even though the Signalwire and PBX codec settings are identical.

***edit to add***

I just noticed that fail2ban is banning 104.248.176.184 and 159.65.244.171...
 
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