DEAL Tried SignalWire.com?

tycho

Guru (not...)
Joined
Aug 9, 2011
Messages
594
Reaction score
229
I have finally been able to receive incoming calls to my Signalwire DID registered to my IncrediblePBX CentOS 6.9, "13.13" instance.

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

I can seemingly ONLY pull this off if I set Allow Anonymous Inbound SIP Calls = yes and Allow SIP Guests = yes. I don't ordinarily ever do that. Is there a way around this that I might be missing? Otherwise, the PBX complains "WARNING,"Rejecting unknown SIP connection from [applicable Signalwire IP address]."" This happens even though I have added all Signalwire-related IP addresses to IPTables.

The PBX will also complain "WARNING[14423][C-00000000] chan_sip.c: Declining non-primary audio stream: audio 16662 RTP/AVP 0 101 13" even though the Signalwire and PBX codec settings are [I believe, and have checked] identical.
Bumping the above post containing a couple of questions (edited in the above quite from the original for brevity and clarity).

These are the IP addresses, obtained after asking in the Slack channel, that I have added to IPTables:

nslookup sip.signalwire.com
Non-authoritative answer:
Name: sip.signalwire.com
Address: 104.248.150.114
Name: sip.signalwire.com
Address: 178.128.235.81
Name: sip.signalwire.com
Address: 188.166.126.7
Name: sip.signalwire.com
Address: 159.65.244.171
Name: sip.signalwire.com
Address: 104.248.176.184
Name: sip.signalwire.com
Address: 167.99.198.84
 
Last edited:
  • Like
Reactions: PBXinmyhouse

PBXinmyhouse

New Member
Joined
Dec 26, 2018
Messages
7
Reaction score
2
Okay, I'm close now... Believe it or not, i got INBOUND working, but outbound still isn't working?!?! Any thoughts?
 

tycho

Guru (not...)
Joined
Aug 9, 2011
Messages
594
Reaction score
229
You've cracked the hard part. As to outbound I've given you all I got in our IM conversation. :) Outbound is the easy one, so double check syntax, look for typos, etc. And make sure you have correctly translated the various long Signalwire strings correctly into Asterisk/FreePBX-speak "name", "host," etc.
 
  • Like
Reactions: PBXinmyhouse

wardmundy

Nerd Uno
Joined
Oct 12, 2007
Messages
15,155
Reaction score
2,632
I would think you could set this up in the same way as CallCentric by adding this block to sip_custom_post.conf assuming you have a signalwire trunk and then adding the IP addresses to your TM3 firewall whitelist:
Code:
[signalwire1](signalwire);
host=sip.signalwire.com

[signalwire2](signalwire);
host=104.248.150.114

[signalwire3](signalwire);
host=178.128.235.81

[signalwire4](signalwire);
host=188.166.126.7

[signalwire5](signalwire);
host=159.65.244.171

[signalwire6](signalwire);
host=104.248.176.184

[signalwire7](signalwire);
host=167.99.198.84
 
Last edited:
  • Like
Reactions: tycho

tycho

Guru (not...)
Joined
Aug 9, 2011
Messages
594
Reaction score
229
That's nicer than what I did. I guess I did it the hard way: I created trunks in IPBX for all of those IPs and added them to the whitelist.

It's actually been working well since I did that. Hmmm; should I mess with success and try your suggestion? Why not -- that's what a sandbox is for!

I never set up Callcentric on my PBX, so I was without that basis for comparison...
 

tycho

Guru (not...)
Joined
Aug 9, 2011
Messages
594
Reaction score
229
Was just playing with my SignalWire setup yesterday and after much initial pulling of hair several months ago it now seems to be working smoothly. Ward, I transitioned to the "Callcentric-esque" sip_custom_post.conf settings that you posted in Reply #24.

Even with 60-second rounding the free credit should last a looooooooong time. A while back I ported a test DID to them and it went smoothly and works well (free porting and a cost of $0.08/month). I may port some long-term but mostly dormant numbers to them as well.

Recommended, once you get past the teething problems.

***

Edit to add: Ward, did you repost that information because the promo code is back in service? I'd read a while back that it had gone defunct...
 
Last edited:

wardmundy

Nerd Uno
Joined
Oct 12, 2007
Messages
15,155
Reaction score
2,632
The SignalWire promo code is a new one. I just modified the old post.
 
  • Like
Reactions: tycho

Frapster

New Member
Joined
Feb 26, 2019
Messages
1
Reaction score
1
The SignalWire promo code is a new one. I just modified the old post.
First off, thanks Ward for all that you do. I've been following you for years and tinkering around but never doing anything really with what I've been reading on your site. I just purchased a hostflyte VPS and am in the midst of setting it up. I also checked out this post and am setting up an account with SignalWire as well. I wanted to report that the code you posted is not working, am I doing something wrong or maybe it expired.

Let me know please. And again, thank you for laying out the knowledge.
 
  • Like
Reactions: wardmundy

wardmundy

Nerd Uno
Joined
Oct 12, 2007
Messages
15,155
Reaction score
2,632
@Frapster: Thanks for the heads up. Looks like they pulled the promo code. We'll post an update if they offer a new one.
 
  • Like
Reactions: tycho

tycho

Guru (not...)
Joined
Aug 9, 2011
Messages
594
Reaction score
229
@Frapster: Thanks for the heads up. Looks like they pulled the promo code. We'll post an update if they offer a new one.
Yes, keep us posted. I forwarded that info to a couple of folks but I'm sure they didn't act on it in time. If a new code comes out I'll give them one more shot and then keep my peace. :)
 
  • Like
Reactions: wardmundy

cfvasquez

New Member
Joined
Jan 26, 2014
Messages
6
Reaction score
0
Can someone post the setting to use signalwire. I can receive calls to asterisk but asterisk is not routing the call properly. From what I see, the incoming call is handled by the outgoing peer in the trunk for some reason. most of the times, when I make or receive a call, there is either domain not found or unauthorized.
 

wardmundy

Nerd Uno
Joined
Oct 12, 2007
Messages
15,155
Reaction score
2,632
Can someone post the setting to use signalwire. I can receive calls to asterisk but asterisk is not routing the call properly. From what I see, the incoming call is handled by the outgoing peer in the trunk for some reason. most of the times, when I make or receive a call, there is either domain not found or unauthorized.
Here are our trunk settings. Inbound Route should be your 11-digit SignalWire DID.

Code:
Trunk Name: signalwire

PEER DETAILS:

type=friend
trustrpid=yes
sendrpid=yes
secret=your-endpoint-secret
qualify=no
insecure=port,invite
host=yourspace-number.sip.signalwire.com
fromuser=7701 ; some existing signalwire endpoint
fromdomain=yourspace-number.sip.signalwire.com
disallow=all
defaultuser=7701 ; some existing signalwire endpoint
context=from-trunk
canreinvite=no
allow=g722&ulaw

Register String: 7701:[email protected]
 

cfvasquez

New Member
Joined
Jan 26, 2014
Messages
6
Reaction score
0
Here are our trunk settings. Inbound Route should be your 11-digit SignalWire DID.

Code:
Trunk Name: signalwire

PEER DETAILS:

type=friend
trustrpid=yes
sendrpid=yes
secret=your-endpoint-secret
qualify=no
insecure=port,invite
host=yourspace-number.sip.signalwire.com
fromuser=7701 ; some existing signalwire endpoint
fromdomain=yourspace-number.sip.signalwire.com
disallow=all
defaultuser=7701 ; some existing signalwire endpoint
context=from-trunk
canreinvite=no
allow=g722&ulaw

Register String: 7701:[email protected]
Hi Ward! Okay to be sure I understand this correctly:

PEER DETAILS are posted above ^^^^ Those are the outgoing settings in the trunk.

Under the incoming settings then the USER CONTEXT:
1. I place my 11 digit signalwire number.
2. I leave the USER DETAILS blank
3. I set my registration string like above ^^^.

Then I go to Inbound routes and I set my DID to my 11 digit signalwire number only.

Finally I go to outbound routes and create an outbound route say signalwire_out and select the signalwire trunk to be used for making calls. Did I miss anything?

raspbx*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
homepbx-991b4cb639d8.sip.signalwire.com Y signalwire 105 Registered Wed, 24 Jul 2019 22:48:38
1 SIP registrations.

I get very weird result. The outgoing call does not connect due to address incomplete.
The incoming connects to my asterisk box but it is not routed properly. It even looks like asterisk trys to dial out to reach a destination. I'm not sure why it initiates a signalwire_out event since that's part of my outgoing routes information.
 

Attachments

wardmundy

Nerd Uno
Joined
Oct 12, 2007
Messages
15,155
Reaction score
2,632
Leave the incoming settings blank. Just enter the registration string.
 

cfvasquez

New Member
Joined
Jan 26, 2014
Messages
6
Reaction score
0
Leave the incoming settings blank. Just enter the registration string.
I keep getting an unauthorized error 401 when my calls get to my freepbx box.


<--- Reliably Transmitting (NAT) to 159.65.244.171:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 159.65.244.171:5060;branch=z9hG4bKd1f7.22a14e0f0b2466497669f0ba5751f87c.0;i=add;received=159.65.244.171;rport=5060
Via: SIP/2.0/TLS 178.128.235.231:35181;received=178.128.235.231;rport=38418;branch=z9hG4bKQ5mgvKUe5cjpS
From: "+1956xxxxxxx" <sip:[email protected]>;tag=BX57pmctj8yrD
To: <sip:[email protected]>;tag=as31aa6eea
Call-ID: 14bab893-2b76-1238-fcbb-0242ac110002
CSeq: 7584054 INVITE
Server: FPBX-14.0.13.4(13.24.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3ca1bf84"
Content-Length: 0

Could it be that the signaling is being blocked somehow?
 

sortons

Member
Joined
Aug 9, 2018
Messages
43
Reaction score
6
@cfvasquez - it looks to me that you have something mixed up in the peer details:
- the 'host=' and 'fromdomain=' are the value to right of '@' in your SIP Endpoint definition in the SignalWire portal
- the 'fromuser=' and 'defaultuser=' are the value to the left of '@' in the SIP Endpoint (the name you chose for your Endpoint)
- the 'secret=' is whatever you type in the 'Change Password' field on the Endpoint page - just type it in again and save before you try registering.

My 'Register string' is '<fromuser>:<secret>:<fromuser>@<host>:5060/+1<DID>'.

In addition to that, follow the suggestions in post #24 - replace 'signalwire' occurences with the name of your trunk. The safest way to add those entries as
indicated is trough 'Admin -> Config Edit -> SIP_custom_post.conf' submit and apply. Then use ./add-ip to whitelist those entries.

On the outgoing route I've prepended a +1/+ for the 10 and respective 11 digits dial patterns.

Then on the 'Settings -> Asterisk SIP Settings -> Chan SIP Settings TAB' make the TLS section look like:
2375
That's about it - you should now have in/outbound calls working.
 
  • Like
Reactions: wardmundy

cfvasquez

New Member
Joined
Jan 26, 2014
Messages
6
Reaction score
0
@cfvasquez - it looks to me that you have something mixed up in the peer details:
- the 'host=' and 'fromdomain=' are the value to right of '@' in your SIP Endpoint definition in the SignalWire portal
- the 'fromuser=' and 'defaultuser=' are the value to the left of '@' in the SIP Endpoint (the name you chose for your Endpoint)
- the 'secret=' is whatever you type in the 'Change Password' field on the Endpoint page - just type it in again and save before you try registering.

My 'Register string' is '<fromuser>:<secret>:<fromuser>@<host>:5060/+1<DID>'.

In addition to that, follow the suggestions in post #24 - replace 'signalwire' occurences with the name of your trunk. The safest way to add those entries as
indicated is trough 'Admin -> Config Edit -> SIP_custom_post.conf' submit and apply. Then use ./add-ip to whitelist those entries.

On the outgoing route I've prepended a +1/+ for the 10 and respective 11 digits dial patterns.

Then on the 'Settings -> Asterisk SIP Settings -> Chan SIP Settings TAB' make the TLS section look like:
View attachment 2375
That's about it - you should now have in/outbound calls working.
It could be a firewall issue, I don't have the ./add-ip script, would someone post a link to add it to my server.

I've done pretty much everything and I still get the 401 unauthorized message.
 

sortons

Member
Joined
Aug 9, 2018
Messages
43
Reaction score
6
I've never seen a recent IPBX install missing the /root/add-ip script. What version of IPBX do you have and on what platform?

Please post the Reports -> Asterisk Info -> Chan_SIP Info, It should be something like this:

Chan_Sip Registry

Host dnsmgr Username Refresh State Reg.Time
gXXXX-XXXXXXXXXXXX.sip.signalwire.com:50 Y vXXXX 105 Registered Sun, 28 Jul 2019 23:18:15
.
Chan_Sip Peers

Name/username Host Dyn Forcerport Comedia ACL Port Status Description
SignalWire/XXXXX 178.128.235.81 Yes Yes 5060 OK (20 ms)
 

Members online

PIAF 5 - Powered by 3CX

Forum statistics

Threads
22,377
Messages
137,433
Members
14,578
Latest member
BoRoDKuH