NEW Tried SignalWire.com?

[billsimon]

There is a simple way to solve the problem where the registration doesn't end up matching a peer and so incoming calls get blocked. If your trunk name is "signalwire" (PEER Name field in your Outgoing trunk settings) then in your registration string replace the host part with just "signalwire"

The registration string would look something like

username:secret@signalwire/DID

 

[Leggomyeggo]

Signalwire still has the new acct signup promotion, REALOGS get you $47 in startup credit to experiment. DID's are only 8 cents a month so if you are careful that $47 could last a long time. Not sure how long promotion will continue, it started 12/10 per a Facebook post.

Credit does not work for existing accts unfortunately.

 

[tycho]

Signalwire still has the new acct signup promotion, REALOGS get you $47 in startup credit to experiment. DID's are only 8 cents a month so if you are careful that $47 could last a long time. Not sure how long promotion will continue, it started 12/10 per a Facebook post.

Credit does not work for existing accts unfortunately.

Even better: create an account as/for a new user. Use the new account signup promo code you mention for the $47. Add your CC info for another $25 credit. "Top-Up" for $5 to turn-off Trial Mode and end up with $77.

 

[Halea]

@tycho : Great suggestion!

@ Everyone in this thread : Thank you for all the useful information. You made my life very easy, it took me less than 30 mn to get an account setup, a phone number verified + a SIP end point created. So far I've made a few domestic calls and they sounded pretty good.
I opened a ticket to turn on international calling as it is off by default on all new accounts apparently. Hopefully international calling will be reliable and with proper CID transfer.

My 2 cents to the configuration discussion : There is no need to purchase a DID if one is only interested in making outbound calls. Just use the registration string without the /DID part at the end. Verify one of your phone numbers so that it can be used as outbound CID otherwise the system makes up bogus caller id numbers.

The cost structure is indeed weird. It looks like signalwire solved the short-calls issue by charging on everything even on unanswered or misdialed calls.

 

[pbxfreeck]

Even better: create an account as/for a new user. Use the new account signup promo code you mention for the $47. Add your CC info for another $25 credit. "Top-Up" for $5 to turn-off Trial Mode and end up with $77.

How can you get another $25 credit? What is CC info?

 

[Halea]

How can you get another $25 credit? What is CC info?

Simply go to their website and create a new account. It will mention that if you also enter your credit card information, only during the initial registration, you will be credited $25. Then on the same screen there is a box for promo codes, enter REALOGS. Finally, press the button to top up your account with $5.00 (which will be drawn from the credit card that you've just entered). So, you will end up with a balance of $77.00, set in permanent mode (not trial) for only $5.00 out of pocket and your credit card info on file with them.

 

[AndyInNYC]

I just signed up and have $77 in credit for my $5 purchase. Thanks!

I verified my phone numbers with SignalWire.

I'd like to use the credit for outbound only.

I'd like to use a PJSIP trunk on my 16-15 PBX - I'm using the defaul 5060 port for PJSIP. All of the directions seem to be for SIP trunk only.

So, at this point, I'm a bit lost.

1. I created a SIP endpoint on SignalWire
2. I created SIP Settings on SignalWire using on Ulaw and Alaw, all the default Ciphers,

Now, I think I need to a) create a PJSIP trunk, b) whitelist each of the IPs c) make calls.

I can get the IP whitelisted, but I'm not sure what settings to use for the PJSIP trunk - help?

BTW, signalwire charged me a penny to verify my 2 phone numbers.

Thanks for the help.

Andrew

 

[Halea]

@AndyInNYC : I used a chan_sip trunk per ward's post further up in this thread, with a small change; I did not use a /DID at the end of the registration string as there will be no incoming calls. I did not white list any of their numbers.

When you create the SIP endpoint on Signalwire make sure that in addition to the user name you create a password (right below the username box). You will use both pieces in your registration string and peer settings with chan_sip.

As for the pennies for the verification calls, you're right. As I said in my post they charge you for everything even for calls that didn't go through and they are not particularly competitive. But 3/4 of a penny per minute rounded to 1 mn is still in the same league as callwithus which charges 0.9 cent/mn with 1mn rounding.

My real interest in Signalwire is to do some digging into their integration capabilities/solutions with freeswitch, so from that venue the bonus starter was great.

Practically, it doesn't hurt to have one more alternate outgoing trunk, especially if I can get good quality international calling, but that is left to be seen as the international call activation is a complicated thing which requires verification (of what I am not sure yet), approval by a their higher-up etc. (In that respect I am already a bit disappointed).

 

[AndyInNYC]

I'll try to figure out a PJSIP trunk.

For the hobbyist, having 128 hours of calling (assuming a penny a minute) for $5 is also a great deal.

Andrew

 

[tycho]

To my way of thinking Signal Wire is a great addition to one's quiver of VoIP providers. It is not trivial to set up but once working seems to chug along. I've been using it for a year now (tip 'O the hat to user @w1ve -- I think he was the one who clued me (all of us?) in initially.

Pure outbound termination comprises 2 charges (all in USD): $0.0065 ("relay call" -- the external transport?) plus $0.0007 ("relay SIP call" the internal processing?) rounded up to a minute so you pay $0.0072/minute with minute rounding. Not the best, not the worst, but the generous credit lets you tinker for quite a while.

Pure inbound shines IMHO. If you let the inbound call stay at SignalWire you pay $0.00255/minute with minute rounding, plus a whopping $0.08/month for a DID. That's better than anything else I use.

I'm still tinkering with using Signalwire's Voicemail hooks; VM works fine via a locally stored LaML bin 'xml file, and can set up to play a remotely stored announcement in your own voice rather than the internal robovoice I'd initially used. I'm not much of a LaML/Perl/Curl/API adept yet so much power is currently over my head. I worked hard to have the leaving of a VM trigger a text to me telling me that I had a VM but was told months ago by one of the devs that I couldn't chain actions like that using internal-only processes and code. Pity, because having used Callcentric and Google Voice and PIAF for so long I rather take VM forwarding for granted. Maybe they have fixed that. But the easy work-around, at a higher cost, is to register the inexpensive Signalwire DID on your own PBX and manage everything there. Or, as I have just tinkered with, have Signalwire forward calls inbound to it's DID to your full-service provider of choice. Forwarding a Signalwire-purchased DID to, say, a Google Voice number is slick and trick, and costs a total of $0.00255 (inbound leg) plus $0.0065, or $0.00905/min. More points of failure, but a cool way to merge the two products for less than a penny a minute.

 

[Halea]

...
Pure inbound shines IMHO. If you let the inbound call stay at SignalWire you pay $0.00255/minute with minute rounding, plus a whopping $0.08/month for a DID. That's better than anything else I use.
...
.

There is better than SignalWire in that respect. I can think of BULKVS, which charges $0.06/month (six cents) per US-48 DID and $0.0003/mn (3/100 of a penny) with 1 second rounding. They port in for free. They have an excellent support group. V1VOIP is not far off with similar pricing but support is not at the same level.
That said I concur with your overall comments on SignalWire. I think it's a good find.

 

[billsimon]

Now, I think I need to a) create a PJSIP trunk, b) whitelist each of the IPs c) make calls.

I can get the IP whitelisted, but I'm not sure what settings to use for the PJSIP trunk - help?

I decided to gamble my $5 on signalwire today, and messed up by not putting in my credit card during the initial setup, and so only got $52 in calling credit. Oh well.

To make calls you do not need to whitelist anything, or register. Whitelist and registration are only for incoming traffic.

I hacked around at this longer than I thought I should have to. I couldn't believe they didn't document their SIP reference anywhere. I bit the bullet and joined their Slack channel.

PJSIP will not connect to their SIP TLS port unless you turn server verification off, and this is because they are using wildcard certs, which PJSIP rejects.

If you are just using SIP over UDP then there is no issue. Really the only reason I wanted to try it out was for the encryption.



advanced tab part 1:


advanced tab part 2 (to use the encryption):

 

[AndyInNYC]

@billsimon - thanks; I'll give it a try this evening (hopefully).

Andrew

 

[tycho]

@billsimon -- so that was you as "Signalwire user" reading them the riot act on the Slack channel today?



"Documented SIP settings?" "We don' need no steenkin' documentation... " Hey, the rest of us muddled through here and on DSLR over a year ago without all that hand-holding </teasing mode OFF>

 

[billsimon]

Yeah, I'm ready to turn in my SIP engineer credential at least for the rest of the year.

I joined earlier and wanted to set up a SIP trunk. Couldn't get TLS to work. Sales robot e-mail comes in and I reply to it saying I'm testing things out, so what are the limitations of the trial mode and where is the SIP documentation? The person who responded said that the trial mode is quite limited (no specifics) and the SIP documentation is at the docs site and if I had any questions please join the Slack channel. There is no actual SIP documentation on the docs site, only REST API info.

I would rather put a nail through my eye than join Slack, so I entered the channel in a foul mood.

Here's the thing. If there's documentation, no one should have to "muddle through," or join a live chat. I see this topic come up over and over. Information shared in a chat but not documented only benefits one person, or a few. I'm not saying give me a step-by-step config. Just a spec sheet.

Now I've got a $50-ish Signalwire calling card to work through in the coming year, which is great. Should last just about forever with the amount of calling I do

Are they disruptive? I dunno. I think Bandwidth, Twilio, Plivo, Telnyx and a bunch of others all have similar APIs and capabilities, and similar pricing. The biggest advantage seems to be simple Freeswitch integration, if you're running that.

 

[Halea]

To make calls you do not need to whitelist anything, or register. Whitelist and registration are only for incoming traffic.

@billsimon : Thank you for your pjsip related post. Regarding your comment about not needing registration to make outbound calls; if the registration info field is left blank on freepbx's trunk settings, how/where would it take the user/password info? Would it be from the peer settings window?

Interestingly, when I set up my trunk, first I had forgotten the registration information, and my calls did not go through. So, my assumption was that the registration string info was being used for the outgoing calls too. But I did many corrections in the peer details window as I cut/paste more than what I should have in first place, so maybe the culprit was elsewhere and got fixed as I corrected my typos.

On a different note, what do you expect the encryption burden to be on the pbx's cpu if it doesn't support AES-NI? Would it be negligible given the audio bandwidth?

 

[AndyInNYC]

@billsimon
I have the PJSIP trunk configured. Unfortunately, the phone on the other end of the call doesn't ring. I'm calling my cell. The PBX thinks the call goes through, but again, no connection to the destination.

I'm sure of my SIP username & password.
The PJSIP trunk shows as registered.

My domain is my 'space url'


Any ideas what I might have wrong?


Andrew

 

[billsimon]

On a different note, what do you expect the encryption burden to be on the pbx's cpu if it doesn't support AES-NI? Would it be negligible given the audio bandwidth?

Maybe a concern if you are on a Pogoplug or pushing a lot of calls. Otherwise I do not think it's anything.

 

[billsimon]

My domain is my 'space url'

It should be the domain of the project, not the space. The project is named like your space then dash then some string of characters randomly selected or you can name it yourself in the project settings. For example: andyinnyc-5fe18a.sip.signalwire.com.

 

[AndyInNYC]

@billsimon

I'm clearly doing something wrong

My signalwire username is ABCD (let's pretend).


My space URL is ABCD.signalwire.com

Under SIP, I created an endpoint which has a username of:
[email protected].
Under SIP Settings I have a SIP uri of
ABCD-[randomstring].sip.signalwire.com

CallerID is set to one of my lines - I'd like to leave it blank to allow my extensions to set push one of my verified phone numbers (... later).
Send As is set to default
Encryption is set to default
Custom Codecs - PMCU, PCMA

In my PJSIP trunk I have
Username: ABCD
Secret: pword as set at signalwire
Authentication set to outbound
Registration is Send
SIP server is sip.signalwire.com
context is from-pstn
Transport is 0.0.0.0-udp (my only choice)

In Advanced:
from domain is ABCD-[randomstring].sip.signalwire.com
from user is ABCD
SRTP is set also


In the cli I see Called PJSIP/[DIALED#]@SignalWire

But the Dialed# doesn't ring. I then see:

[2019-12-24 12:51:58] WARNING[3600]: res_pjsip_outbound_registration.c:792 schedule_retry: Temporal response '401' received from 'sip:sip.signalwire.com' on registration attempt to 'sip:[email protected]', retrying in '60'


Since I only have 5 settings or so at signalwire and 4-5 in the trunk settings, I can't figure out what I have wrong. No obvious typos, though.

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


Andrew