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Account Settings Server Error

March 19, 2019, 12:10 pm
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@bburd wrote:

Getting the following error on my Sipstation module when i try to refresh asterisk account information.

“An error occurred trying to contact the server for account settings.”

This was setup and working flawlessly (for months) until i did one of three things,

#1 Upgrading Sipstaion module to 13.0.15
#2 Removed services (canceled lines) on one of the two sipstation accounts
#3 Reset my SIPStation SIP password (as per the request sent to my email)

Please let me know if there is anything else I can post to help you help me!

Thank you for your time!

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Sipstation SMS

March 21, 2019, 6:25 am
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@lzcantrell wrote:

What is the determining factor for aDID to be SMS capable, when porting into Sipstation? I ported a number from ATT and is not SMS capable now. I inquired to Sangoma Support before submitting the the port, and the response was…Yes it would be SMS capable, and only in rare cases a DID would not be SMS capable.

Thanks
Luke

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Sipstation module "The server is currently not responding"

April 11, 2019, 9:47 am
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@SterlingPkg wrote:

My Sipstation trunk is working correctly, however I’ve noticed the last two days this error message appears in the SIPSTATION module.
The only change to the system recently was update roll-out last week, where the module was updated to 14.0.1.8

Not panicking, since obviously calls are being routed fine, but still concerning.
Is this something on my end, or should I kick this up to support?

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SIPstation server unavailable error

April 19, 2019, 8:29 am
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Trunk SIP incoming doesnt work

May 13, 2019, 7:48 am
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Sip trunking with 2 router(sip trunk) with 1 router(internet)

June 17, 2019, 10:09 am
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@Ezra5447 wrote:

Hi Guys,

Currently our service provider installed 2 set of ONU modem which is responsible for sip trunk. each modem has wan ip to add it as trunk host (eg . 10.xx.xx.xx) . on the other hand, we have another router which used for Internet purposes which used by extension user , freepbx box and other servers (192.168.1.xx). Now because of the 2 set of sip trunk modem and the internet modem is different network. how can i make the freepbx box and 2 sip trunk router to listen each other.
I have searched and found that i can configure SWITCH by creating 2 vlans and connect them together. is it possible that way? or any other better suggestion i could get?. Thanks

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SipStation trunks timing out and not registering

July 6, 2019, 11:35 am
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@brisbinj wrote:

I am running the latest distro after having an issue where my sipstation trunks are not registering.

This is the error i get from the CLI in asterisk

[2019-07-06 18:32:35] NOTICE[29591]: chan_sip.c:15976 sip_reg_timeout: – Registration for ‘7p36NixOaZNq@trunk2.freepbx.com’ timed out, trying again (Attempt #21)

I just did a new clean install of FreePBX Distro on my machine this morning and entered the key for my sipstation trunks and still having the same issue.

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SIPStation issues

August 14, 2019, 12:43 pm
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@Benn5325 wrote:

Somewhat lost here, totally new to SIPstation… We had to get some new DIDs for the Phoenix area, and as Comcast for some reason cannot supply them we thought we’d try SIPStation.

Quick rundown of our setup.
FreePBX 14.0.13.4 Asterisk 16.3.0
We have 1 SIP trunk with Comcast and have a Adtran supplied by them.
We also use a Sophos firewall and control all the traffic from there.

I have 2 issues…
1… No audio either way. I have one of the DIDs point to my extension. When I call from my cell it rings and I can pick up, just no audio…
2. SIPStation module is showing a error that the server is currently unavailable and cannot process my request. It is up because I can open the Sipstation Store…

Because we are going through the adtran, my external ip in SIP settings is a local IP to the adtran.
All other calls and audio are fine, we have over 200 endpoints on the system.
It is only this hand full of DIDs from SIPStation.

I know it’s something with the IPs, just need some pointers please.
Thanks

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Sipstation and Vega

September 19, 2019, 10:56 am
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Trunks not registering

October 2, 2019, 12:57 pm
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@arjones5 wrote:

I have an active ticket with SIPStation for this issue but I’m desperate and we’re not making any headway. I was getting service issues where the trunk would just throw me a SIP Ping Unreachable error, now it is constantly not able to register the trunk. The DNS request is making it out, my ISP isn’t blocking the protocol, I just can’t seem to get any traction on the issue - let along determining what the root cause it.

[2019-10-02 14:49:00] NOTICE[1362] chan_sip.c: – Registration for ‘[user]@trunk1.freepbx.com’ timed out, trying again (Attempt #20)
[2019-10-02 14:49:01] NOTICE[1362] chan_sip.c: – Registration for ‘[user]@trunk2.freepbx.com’ timed out, trying again (Attempt #20)

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911 Going to Wrong PSAP

October 16, 2019, 11:49 am
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@MollyMae wrote:

Greetings,
I have an issue and I’m not sure where to address the concern. It’s probably not FreePBX (I believe it’s the SIP provider), but I’m hoping I can confirm that before reaching out.

I recently configured E911 CIDs for a company with 5 locations. The CID information (including the address) is passed correctly to the dispatcher, but when I dial 911 it goes to the county dispatch instead of the city dispatch. I confirmed with the county dispatch that based on the address, it should be going to the city.

I use SIPStation, so I will be reaching out to them next unless someone knows what else could be the issue. I don’t know if any config information would help, since I think FreePBX is passing out the Emergency Route correctly, but I’ll gladly provide any information requested.

Thanks,
Molly

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SIPStation SMS missing from UCP tab userman > edit user > UCP tab >

October 24, 2019, 1:59 am
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No voice/audio on inbound/outbound calls

October 24, 2019, 10:42 am
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@arjones5 wrote:

After a recent firewall upgrade, I cannot get audio for any outbound or inbound phone calls. Trunking is working, I believe it’s something with the SIP handshake that’s getting lost. Looking at the debug logs, nothing jumps out at me as an issue. I’ve read through the other posts to no avail. I’m pretty sure this is a network issue, just not sure where to start.

Here are the odd balls…

  • If an inbound call is invited and acknowledged, I can hear the internal phone from the external phone but the external phone does not transmit internally
    *Rebooting the PBX does nothing
    *The new firewall has packet inspection active whereas the old “firewall” was a Netgear Nighthawk. Tried turning it off and there are no changes.
    *Firewall ports are open for 5160 to the PBX from the trunk IPs
    *SIPSTATION is the trunking provider

I’m scratching my head trying to hunt down where this dies.

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Faxstation ports How to Separate?

November 20, 2019, 11:00 am
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@shackbill wrote:

So i have 1 Fax trunk and 2 DIDs. I am using ports 1 and 2 for each of 2 faxes. But on the ‘Fax Trunk to Device Assignments’ I can only drag 2 DIDs into 1 port? All outgoing faxes appear to be from one DID. Any advice how to properly config this?

Also, i have a few faxes that are sent where the state says ‘SUCCESSFUL’ but the receiver says they never got it. Is there a way to trace that?

SeparatePorts
SeparatePorts.PNG1031×643 34.6 KB

Im Guessing i should be able to set this up for 2 ports - 1 did for each? But i dont see how.

FaxTrunks
FaxTrunks.PNG1074×368 33.2 KB

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How to analyze my trunk usage

December 5, 2019, 8:37 am
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@benreisner1 wrote:

I’m looking to try to analyze my SIPSTATION trunk usage so that I buy the optimum number of flat rate trunks. Is there anything you can suggest for this?

For example: Currently I have 3 Flat Rate Trunks, and I have bursting enabled so that if I sometimes use more then 3 chanels it will just charge me per-minute for them. I know that I sometimes burst above 3 trunks.

I’m trying to analyze so that I can purchase the optimum number of trunks.

If I could generate some report either from SIPSTATION or from my FreePBX box that would say for a given time period how many minutes were at 0 simultaneous calls, how many were at 1, how many were at 2, 3, 4, 5, 6, etc… I’m getting somewhat close to success using the CDR records but I feel like I have to be re-inventing the wheel

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Outbound call gives fast busy, after hanging up, the dialed phone receives a call - both internal and external dialing

December 11, 2019, 6:36 pm
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@jeanaf wrote:

Today I installed SIPStation trunks, in my migration from another SIP provider to SIPStation. As soon as I added the key, and the routes and trunks were created, my outbound calls started getting fast busy. But the really weird thing is, when we hang up the call, about 10 seconds later the dialed phone rings. I have tried to remove the trunks, and go back to my other still working trunk, all to no avail. I think I might have a problem with my dial plan, but I don’t know how to troubleshoot that. Any help or thoughts would be greatly appreciated!

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All circuits are busy now

January 2, 2020, 9:06 am
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@zablith wrote:

Hello,

While everything was working fine for the previous months, suddendly we are unable to send or receive external calls
while trying to dial external line, we receive the following message:
All circuit are busy now.
can’t figure out what’s happening.

Please help

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SIPStation Major Outage

January 20, 2020, 5:13 pm
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@mmoo9154 wrote:

Is anyone else seeing that SIPStation is having a major outage. Our trunks are down, and the 920-886-8130 number reports that no one is there to take support calls.

The status.sangoma.com page reports numerous “Majpr Outage” but says that SIPStation - Trunking is Operational.

Unfortunately, none of our incoming numbers are working.

Anyone know what’s going on??

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Sipstation customer offline

January 20, 2020, 11:05 pm
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@incognito wrote:

My customer has been offline for the past 6+ hours. We had setup a failover phone service provider but that is not currently happening. My customer’s phone are just dead… There is no way for us to failover or forward to anywhere else.

This is only affecting our most recent client that signed up with Sipstation. They are ready to jump ship if things are not working soon :frowning: . Is there a reason that not even our basic failover DID is working?

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2 sipstation accounts on one pbx

February 1, 2020, 10:00 pm
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@bajramia wrote:

I have a customer who has to different companies on same office with one pbxact, he wants 2 different sipstation accounts one for company A another one for company B. My question is can i create 2 sipstation accounts and configure on one pbxact

Thank you
All

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