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SIPStation Call Forward

December 21, 2022, 9:39 am
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Hello.

A winter storm is threatening to cut off power to my business later this week. We use FreePBX and SIPStation. I have the failover set an in house cell phone. FreePBX is configured to send all call received when we are closed to our emergency service. My concern is if we lose power in the middle of the night, SIPStation will forward our calls to the cell phone instead of our emergency service.

When we used PSTN, we dialed *72 and then the number to call forward at the phone company. Is there a similar sequence we can use to forward our calls at SIPStation (before they are sent to our FreePBX server that may or may no be online due to power problems)?

Can I set this up as a Feature Code to simplify implementation?

Thank you very much for looking through this and considering my questions.

drnate

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Does SipStation offer encryption media and signaling?

February 5, 2023, 10:32 am
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I haven’t been able to find a definitive answer. Does the Sangoma SipStation sip trunks offer TLS encryption for both the media and the signaling?

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No Sounds with SIP Station, but register OK and Firewall OK

February 21, 2023, 11:34 am
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I am thinking of moving my trunks over to SIPStation. I signed for for a trial and have run through the setup in the sipstation module. The SIPstation trunks register OK and I can make an inbound call that I can see is being routed through FreePBX even to a follow-me. However, I can’t get ANY sound incoming or outgoing what so ever.

Again, the firewall test shows as good. However, the Your Contact IP and Your Network IP are showing as yellow. Note: My Freepbx install is on a local IP, routing through PFsense.

Note: I do get this error in the SIPStation interface: “Chan_PJSIP is not supported when SIP Driver is forced to Chan_SIP”

Any suggestions. The WIKI is broken, so am having a hard time finding documentation.

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VoipInnovations 404 err, no route found

April 21, 2023, 9:50 am
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Identifying details anonymized. Dialing muliple 10 digit North American numbers.

I don’t think it’s the PBX since nothing was changed on it as far as I know. Previous SIP messages look like normal call negotiation until this one. The end result is an all circuits busy message.

2023/04/21 11:33:24.099039 64.136.174.30:5060 -> xx.xx.xx.xx:5060
SIP/2.0 404 ERS:No Route Available
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK34c063f5;rport=5060
To: <sip:NXXNXXXXXX@64.136.174.30>;tag=sansay845362775rdb28095
From: "test" <sip:XXX@xx.xx.xx.xx>;tag=as71b725e8
Call-ID: 6bb53879430b98251b7be52407a5dcd7@xx.xx.xx.xx:5060
CSeq: 102 INVITE
Content-Length: 0

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VOIP Innovations Cloud Network

April 30, 2023, 8:11 pm
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Good Morning Colleagues
I hope you are doing fine . I was checked Firewall ==> Networks and i have found the below entries that i didn’t as a local zone . Does my system is hacked ?

Voip Innovations Cloud Network

Best Regards

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Sipstation Connectivity Check Fails

May 14, 2023, 5:29 am
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When performing the Connectivity Check in the SIPstation module I get the following result:

image

IPs for external are configured correctly. Port forwarding for SIPstation is configured correctly. Equipment and Configuration is the same as previous FreePBX and SIPstation installs. I have 2 way audio on inbound and outbound calls. However still I get this failure when running the connectivity test.

I see where someone posted that opening mirror.freepbx.com would solve this, but we have not had to do that in the past. Also mirror.freepbx.com resolves to multiple IP addresses which would mean multiple firewall rules to add just for the test to pass when the turnks already work, which doesn’t make a lot of sense.

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SIPstation External Connectivity check is timing out

May 25, 2023, 6:33 am
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This is a new installation with the 20-day trial.

The Check Connectivity is showing the Firewall Status as Fail and pops up a banner message: The test timed out which means your firewall is probably configured wrong. If subsequent tests fail, check your port forwarding on the firewall.

The other anomaly is the message: The current trunks are utilizing chan_pjsip and connecting to trunktrial1.freepbx.com & trunktrial2.freepbx.com over port Unknown using an RTP range of 10000-20000

What is the Unknown port? I am wondering if I missed something with the FreePBX configuration?

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VI Module configuration or just setup IP configuration

July 5, 2023, 2:02 pm
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We are moving back to VoIP Innovations. Was going to use the FreePBX VI module for configurating VI endpoints mapped to FreePBX instances. Using the module means all DIDs have to have inbound CNAM. We tend to have lots of clients that have many DIDs that are not used. We have one client with 300 DIDs and only 25 have traffic - but they want the “unused” DIDs because they are contiguous. Having these unused DIDs now cost us an extra $.55 / DID / month for inbound CNAM that is not being used. This one client alone would cost us $150 / month. We have several like this.

If I configure Endpoints to IP authentication in VI, rather than use the module, I know that SMS will not work. Are there other considerations I am missing that make the VI Module a more feature rich experience versus only doing IP based authentication? Will more integrations depend on the module in the future?

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Sipstation

August 14, 2023, 6:37 pm
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I bought two sip trunks in Sipstation, but wish to disable registration and allow traffic from a set of IPs.
I added the second IP in the Sipstation store, but it allows traffic only from Free-PBX, not from my IP that does not use registration.
Is this model even possible?

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SipStation configuration

September 1, 2023, 9:38 pm
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I’ve just moved my FreePBX server to a new network with a new static IP address and am unable to get it to accept calls. I’ve double checked my firewall and I can’t find anything wrong with the port forwarding.

I’ve added the new IP Address to the SIPSTATION Notification and Access Control section of the trunk group ACL settings.

I’ve added the new IP address to the Asterisk SIP settings in both the General SIP Settings Tab
and in the SIP Legacy settings tab.

In the SIPstation module shows:

Asterisk Registration Status:
Request Sent

Your Contact IP: Not Available (in grey)
Your Network IP: Not Available (in grey)
SIP Ping: OK (in green)

External connectivity check shows no firewall problems and my correct external IP address

I know that I can call support, but they are closed for the next 3 days and I really need to get the phones up before the end of the long weekend if at all possible. I think I’m missing something obvious. Any help would be greatly appreciated.

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SMS forwarding or failing over?

September 5, 2023, 12:27 pm
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I had a text end up at my cell phone yesterday morning which is set at Voip Innovations as the failover for that particular DID. I dont see anything in the asterisk log or in UCP and VI support swears that there is no such thing as an SMS failover. They suggested I talk to Sangoma about it. They do acknowledge though that the same exact message came in from a number I dont recognize to a DID in the PBX and then 2 seconds later was sent out to my cell phone from the DID.

I also checked the 10DLC registration for that DID and my cell phone number doesnt appear anywhere in there.

Anyone at Sangoma familiar enough with both sides of this to know how to diagnose this?

Also, is this feature something built into the PBX somewhere?

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Incoming Sipstation sms issue

December 6, 2023, 7:26 am
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I’m setting up freepbx with the sipstation sms service using UCP as an interface

Initially I was able to receive text messages into the ucp, however that stopped, and I’ve been trying to figure out if my config is broken. Unfortunately all of the sms setup pages from the wiki are down from a wiki overhaul from what I can tell, any links I can find are broken and searches find those same broken links

When I send a text out from ucp it works perfectly, when I send one in I get this in my asterisk logs, and nothing at all in the ucp, replacing number for the did the message was sent to

pbx.c: Executing [number@sms-incoming:1] NoOp(“Message/ast_msg_queue”, “SMS came in with DID: number”) in new stack

pbx.c: Executing [number@sms-incoming:2] Goto(“Message/ast_msg_queue”, “s,1”) in new stack

pbx_builtins.c: Goto (sms-incoming,s,1)

pbx.c: Executing [s@sms-incoming:1] AGI(“Message/ast_msg_queue”, “agi://127.0.0.1/sipstation_sms.php, RECEIVE”) in new stack

res_agi.c: agi://127.0.0.1/sipstation_sms.php, RECEIVE: SMS RESULT: SUCCESS

res_agi.c: <Message/ast_msg_queue>AGI Script agi://127.0.0.1/sipstation_sms.php completed, returning 0

pbx.c: Executing [s@sms-incoming:2] Hangup(“Message/ast_msg_queue”, “”) in new stack

pbx.c: Spawn extension (sms-incoming, s, 2) exited non-zero on ‘Message/ast_msg_queue’

Firewall on the pbx is off, external hardware firewall isnt blocking anything in logs, it appears the message is getting to the pbx just fine, just not making it to the ucp

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SIPStation support...now at 65 minutes on hold

January 3, 2024, 1:02 pm
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Trunks are down. Started at #2 in the SIPStation support queue and after 65 minutes am at #1.

This has to work better…I’m at risk of getting to the 2:00 PM Pacific Time close of business with non-functioning trunks.

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All Circuits are Busy Now SIPStation

January 13, 2024, 1:09 pm
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Hello,

I have gotten my sip station trial but when I make an outbound call I get the error All circuits are busy now.

In the trunk I do see this error:

image

However I have routes like SIPStation-Out and OUTbound (custom configured) like this:

image

with more

But I get the error even when dialing correctly

image

PLease help, thnks

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Unable to receive texts

February 12, 2024, 9:06 am
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On Friday, we were able to receive texts for MFA reasons but since Saturday we are unable to receive any texts but can send them. These are the updates that auto updated on Saturday morning (manager 16.0.21 (current: 16.0.20) restapps 16.0.36.5 (current: 16.0.36.4) sangomaconnect 16.0.45.2 (current: 16.0.44.38) sangomartapi 16.0.47.1 (current: 16.0.46.65) sysadmin 16.0.41.20 (current: 16.0.41.18)).

I tried to roll them back but unable to do so. i uninstalled the sangomaconnect and the sangomrtapi module but same result. These numbers were bought via SIPStation, I called their support line and was placed in a callback queue 2.5 hours ago. Anyone run into this?

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No inbound SMS

April 26, 2024, 8:12 am
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Using latest stable version of voipinnovations module and latest stable version of asterisk 18. Outbound SMS works but no inbound. I allowed all in on firewall to makes sure that is not blocking.

Where should I be looking for the incoming SMS on the eth0 interface? Is it an API and if so what port? Or is it just a UDP packet on my SIP port I can look for with sngrep? According to this article it looks like it is sent SMS as SIP, so I should see it in sngrep but I don’t.
https://voipinnovations.atlassian.net/wiki/spaces/VIW/pages/588382473/SMS+on+DIDs

How is SMS webhook used with VIModule if at all? I couldn’t seem to find any info on that anywhere.

Other than UCP, do I need to use sangoma phones/software to receive SMS from VI module on a phone/app or will any SIP softphone capable of receiving standard SIP messages (like MicroSIP) work?

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SIPStation Down?

May 29, 2024, 1:41 pm
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Does anyone use SIPStation and, if so, is it up for them? I’m trying to get around a faxing issue with our main PBX by using FreePBX for HR to be able to receive faxes at. However, inbound calling isn’t working in FreePBX and SIPStation page under Connectivity shows this:

image

I rebooted the server and still nothing. I should add that we still pay for SIPStation, just got a receipt yesterday from them for our monthly dues.

Just trying to get a quick workaround, was hoping this would work. Only a week and a half of school left where I am, so hopefully our main PBX support gets fax working again for that and I won’t have to worry about using FreePBX right now.

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HangUp CAUSE

July 8, 2024, 4:44 pm
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Hello…

We have a FREEPBX 15 (15.0.37.5). hosting an DID INTL for MEX (through VoipInnovations)

we are experiencing some issues while assigning call to extension, as in a random behavior, we have got :
“HANGUP CAUSE: 0” or “HANGUP CAUSE: 127”

This happens with the same caller… some time the call is accepted and sometime is dropped … for a call coming from the same origin.

End user states the call rings but unable to take the call.

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Unable to receive calls

September 24, 2024, 10:53 am
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I’ve been testing out SIPStation with FreePBX 16, and I’ve been able to make calls but not receive them. Phone tries to connect and then disconnects. Nothing even appears on the SIPStation account. Sangoma support tells me that there are two IP addresses that they’re seeing. The *.206 address is the main company address, and the other is a static NAT assignment that FreePBX should be using. All traffic should be NATted with *.213, so how do I change the Contact IP?

Screenshot 2024-09-24 124206

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Multiple SMS recipients

October 17, 2024, 8:14 am
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  1. Is there any way to send out a text to multiple recipients using SIPStation?

  2. Is there a was to send an automated SMS through a script?

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