@mmoo9154 wrote:

We just had a a strange interruption to out service. Everything has been working great. There have been no updates or changes at all to FreePBX/Asterisk. All the endpoints are working, registered (have service), and can place calls internally.

Unfortunately and out of the blue, we started getting a strange “Q.850; cause=17” message whenever we tried to dial an out side line. (FWIW, all our phones are Yealink, mostly T27G and T48G.)

At the same time we started getting complaints (over cell and email) that people cannot call in as well.

I do not see anything wrong on the FreePBX Dashboard, nor on the SIP provider’s status board. So, I rebooted our server just to se if that would clear the issue.

It didn’t. After a long but regular reboot, all of the endpoints successfully (re)registered, and for a minute or so I could still get the Q.850; cause=17 error.

Now, the Q.850 message does not show up anymore, but when trying to dial out, it emadiately goes to the “All circuits are busy now. Please try your call again later” voice message. And, when trying to dial in on any of the DIDs, it just give busy signal.

Does anyone have any ideas on what logs I can look at or where else I might spelunk in oredr to find what’s failing?

Every things looks just fine, except it isn’t.

Thanks!

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@lcs226 wrote:

Trying SIPSTATION free trial to demonstrate for customer but cannot make or accept call. Email said that it was setup automatically in my freePBX. Not sure how to check what setting might be wrong. Any Ideas
please?

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@chuckjuhl wrote:

Calls are being dropped after 30 minutes. This happens when the employee is talking. See attached. There are no session timers in our FreePBX configuration that I can find. Our employees often have long calls when dealing with patients as well as insurance providers. We are using sipstation unlimited channels with cyberlink/Sangoma cloud hosting (freepbxhosting.com), Free PBX 14, Digium D70 phones.

The configuration is identical to the configuration at another location (but different SIPStation and FreePBXhosting accounts). The second location does not have this issue. The only difference between the two locations is that at the location referenced here call recordings are set to “Force” on inbound and outbound routes while call recording at the second location not having the call limit issue is set to “Don’t Care.”

Is there a time limit for recording a call? If so, is there a way to modify or override that limit?

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@NevadaTech wrote:

Hello,

I have a clean install of FreePBX 2002.02 It’s working fine except for sening/receiving calls. Mainly, I’m trying use SIPstation SIP service. When I look at the Connectivity> SIPstation I get:

WARNING The server is currently not responding. It is either unavailable or access is being blocked. If the server is unavailable, please try again later.

I submitted a ticket to SIPstation and received this response:

Your firewall is not allowing traffic back from push2.schmoozecomcom and that is what is required for the module to show proper status. Currently I show you registered to trunk1 but there is a problem because the port is being translated. You need to set your firewall to not translate the port of traffic from your PBX to our trunking servers.

So, how do I fix this? Some of my testing has been to place the server on the Internet with its own public IP (same error) and the FreePBX firewall enable; I’ve built a clean PBX and placed it behind a pfSense router in it’s own DMZ and open the ports (5060 TCP/UDP and 10000-20000 UDP) + tried 1:1 NAT + tried port forwarding (same error).

Initially I though it was a pfSense issue but putting it flat on the Internet excluded that culprit. I don’t see in the FreePBX Firewall how to set ports or NAT’ing. Using the GUI first and then fwconsole to make sure trunk1/trunk2.freepbxcom and push2.schmoozecomcom were Trusted and exclude from firewall does not change the error message. FYI, I know the trusted should be .com but this generates a forum pasting error.

If I delete the key the SIPstation module comes the page renders properly, until I re-add my key.

So what obvious thing am I missing? Thanks!

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@stasul wrote:

Hi there!
We are trying to demonstrate a SIP call over freePBX using a SIP soft client on one end and a regular mobile or landline phone on the other. Installed FreePBX 15.0.16.49 using distribution package in a Linux VM. We can make calls between two SIP soft client extensions after we opened up the firewall to pass SIP and RTP traffic. Also enabled internet access from PBX VM on the firewall (we were able to download and install module updates - currently all latest modules). The next step was to enable SIPSTATION free trial to set up a SIP trunk and demonstrate the call to/from a mobile phone. However, when I go to Connectivity->SIPSTATION menu, the “SIPSTATION Store” and “SIPSTATION Free Trial” tabs show “cannot reach the page” message and a red banner pops up on top with “undefined” error. Any suggestions on what I may be missing? Thanks!

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@jheisler99 wrote:

The credit card I had on file for the SIPstation store expired so I had to put in a new one. After I put in a new one, it charged the card. But I have been trying to contact the support team because I think they need to turn it back on for my company but they have been closed. Does anyone know when they open?

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@gberg wrote:

When i’m trying to start my trial via the web GUI it gives my a red error bar in the top of the screen. This happens when im going from page 1 (country data) to page 2 after the red bar disappears an infinite loading screen pops up

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@bobconfino wrote:

Greetings,
I am trying to determine requirements to configure our Cisco ASA for success with our future PBXAct60 (192.168.115.30) and SIP Station trunking. The Sangoma Wiki and other docs are all over the place concerning support access, SIP/RTP access, and SIP Station access. I would appreciate a sanity review of my current understanding of the internal/external network paths below. (All of our auto provisioning nodes are on our internal network).

Any constructive comments and references to authorative documents welcomed.

Best Regards,
Bob Confino - Volunteer Tech Team - New Life Bible Fellowship Church

?Sangoma Support access requirements?

*Port forward traffic (NAT) from Sangoma support to the VoIP PBX

Remote console access (ssh) for Sangoma Support
object network on-PBXAct-tcp-22
host 192.168.115.30
nat (inside2,outside) static interface service tcp 22 22

Remote OpenVPN access for Sangoma Support
object network on-PBXAct-udp-1194
host 192.168.115.30
nat (inside2,outside) static interface service udp 1194 1194

Remote OpenVPN access for Sangoma Support
object network on-PBXAct-tcp-1194
host 192.168.115.30
nat (inside2,outside) static interface service tcp 1194 1194

?Sangoma SIP traffic requirements?

*Port forward traffic (NAT) from the telephony ISP to the VoIP PBX

SIP traffic from Sangoma SIP Station Service (Telephony ISP) (SIP Trunking)
object network on-PBXAct-udp-5060
host 192.168.115.30
nat (inside2,outside) static interface service udp 5060 5060

SIP traffic from Sangoma SIP Station Service (Telephony ISP) (SIP Trunking)
object network on-PBXAct-udp-5061
host 192.168.115.30
nat (inside2,outside) static interface service udp 5061 5061

*ACLs for support traffic and SIP Station traffic
<Note:must remove ‘any’ from ACEs with ITSP IP address>
access-list outside_in permit tcp any host 192.168.115.30 eq 22 (enable/disable as needed)
access-list outside_in permit tcp any host 192.168.115.30 eq 1194 (enable/disable as needed)
access-list outside_in permit udp any host 192.168.115.30 eq 1194 (enable/disable as needed)
<Note:must remove ‘any’ from ACEs with ITSP IP address>
access-list outside_in permit udp any host 192.168.115.30 eq 5060
access-list outside_in permit udp any host 192.168.115.30 eq 5061

?Sangoma RTP traffic requirements?

**Port forward range of ports for RTP traffic from telephony ISP to the VoIP PBX

object network on-PBXAct60
host 192.168.115.30

object service os-udp-RTP-Ports-Range
service udp destination range 10000 11000

nat (inside2,outside) static interface service os-udp-RTP-Ports-Range os-udp-RTP-Ports-Range

**ACLs for RTP traffic
access-list outside_in extended permit udp any host 192.168.115.30 range 10000 11000
<must remove ‘any’ from above ACL with ITSP IP address>

Sangoma Node List
trunk1.freepbx.com 192.159.66.3
trunk2.freepbx.com 162.253.134.142
trunktrial1.freepbx.com 162.253.134.135
trunktrial2.freepbx.com 192.159.66.4
push2.schmoozecom.com 199.102.239.11

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@jmbldwn wrote:

I’ve set up a FreePBX instance, and I’m testing it with the SIPStation trunk free trial.

I am able to connect SIP phones to it and they seem to register correctly.

I am able to successfully call a SIP phone from the PSTN, so inbound calling is working.

But I am not able to make outbound calls. I am getting an “All circuits are busy” message when I try to call from a SIP phone.

There isn’t much in /var/log/asterisk/full, but /var/log/asterisk/core-fastagi_out.log shows some errors (below).

Any suggestions on how to debug this?

2020-10-14 00:19 +00:00: [54532] Asterisk connection opened
[54532] Launching ./sangomacrm.agi with args:
2020-10-14 00:19 +00:00: [54532][1602634767.9] >>> agi_network: yes
[54532][1602634767.9] >>> agi_network_script: sangomacrm.agi
[54532][1602634767.9] >>> agi_request: agi
[54532][1602634767.9] >>> agi_channel: PJSIP/3-00000003
[54532][1602634767.9] >>> agi_language: en
[54532][1602634767.9] >>> agi_type: PJSIP
[54532][1602634767.9] >>> agi_uniqueid: 1602634767.9
[54532][1602634767.9] >>> agi_version: 16.11.1
[54532][1602634767.9] >>> agi_callerid: 6666
[54532][1602634767.9] >>> agi_calleridname: unknown
[54532][1602634767.9] >>> agi_callingpres: 0
[54532][1602634767.9] >>> agi_callingani2: 0
[54532][1602634767.9] >>> agi_callington: 0
[54532][1602634767.9] >>> agi_callingtns: 0
[54532][1602634767.9] >>> agi_dnid: 16502454652
[54532][1602634767.9] >>> agi_rdnis: unknown
[54532][1602634767.9] >>> agi_context: macro-dialout-trunk
[54532][1602634767.9] >>> agi_extension: s
[54532][1602634767.9] >>> agi_priority: 24
[54532][1602634767.9] >>> agi_enhanced: 0.0
[54532][1602634767.9] >>> agi_accountcode:
[54532][1602634767.9] >>> agi_threadid: 140682880235264
[54532][1602634767.9] >>> agi_ASTAGIDIR: /var/lib/asterisk/agi-bin
[54532][1602634767.9] >>> agi_port: 54532
[54532][1602634767.9] >>>
2020-10-14 00:19 +00:00: [54532][1602634767.9] >>> {“code”:“ECONNRESET”,“errno”:“ECONNRESET”,“syscall”:“read”}
2020-10-14 00:19 +00:00: [54532] Script ended with code 1 and signal null
2020-10-14 00:19 +00:00: [54532][1602634767.9] >>TRIED TO SEND TO DEAD AGI>> HANGUP
2020-10-14 00:19 +00:00: [54532] Asterisk connection closed
2020-10-14 00:19 +00:00: [54540] Asterisk connection opened
2020-10-14 00:19 +00:00: [54540] Launching ./sangomacrm.agi with args:
2020-10-14 00:19 +00:00: [54540][1602634767.9] >>> agi_network: yes
2020-10-14 00:19 +00:00: [54540][1602634767.9] >>> agi_network_script: sangomacrm.agi
2020-10-14 00:19 +00:00: [54540][1602634767.9] >>> agi_request: agi
2020-10-14 00:19 +00:00: [54540][1602634767.9] >>> agi_channel: PJSIP/3-00000003
[54540][1602634767.9] >>> agi_language: en
[54540][1602634767.9] >>> agi_type: PJSIP
[54540][1602634767.9] >>> agi_uniqueid: 1602634767.9
[54540][1602634767.9] >>> agi_version: 16.11.1
2020-10-14 00:19 +00:00: [54540][1602634767.9] >>> agi_callerid: 3
[54540][1602634767.9] >>> agi_calleridname: unknown
[54540][1602634767.9] >>> agi_callingpres: 0
2020-10-14 00:19 +00:00: [54540][1602634767.9] >>> agi_callingani2: 0
[54540][1602634767.9] >>> agi_callington: 0
[54540][1602634767.9] >>> agi_callingtns: 0
[54540][1602634767.9] >>> agi_dnid: 16502454652
2020-10-14 00:19 +00:00: [54540][1602634767.9] >>> agi_rdnis: unknown
[54540][1602634767.9] >>> agi_context: crm-hangup
[54540][1602634767.9] >>> agi_extension: s
2020-10-14 00:19 +00:00: [54540][1602634767.9] >>> agi_priority: 7
[54540][1602634767.9] >>> agi_enhanced: 0.0
[54540][1602634767.9] >>> agi_accountcode:
2020-10-14 00:19 +00:00: [54540][1602634767.9] >>> agi_threadid: 140682880235264
[54540][1602634767.9] >>> agi_ASTAGIDIR: /var/lib/asterisk/agi-bin
2020-10-14 00:19 +00:00: [54540][1602634767.9] >>> agi_port: 54540
[54540][1602634767.9] >>>
2020-10-14 00:19 +00:00: [54540][1602634767.9] >>> {“code”:“ECONNRESET”,“errno”:“ECONNRESET”,“syscall”:“read”}
2020-10-14 00:19 +00:00: [54540] Script ended with code 1 and signal null
2020-10-14 00:19 +00:00: [54540][1602634767.9] >>TRIED TO SEND TO DEAD AGI>> HANGUP
2020-10-14 00:19 +00:00: [54540] Asterisk connection closed

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@Bradbpw wrote:

Is there any way to get email alerts when a SMS is received through SIPStation SMS? I know I can retrieve them on UCP but I’d prefer to not have an extra place for my employees to look for messages. I can deal with them going to UCP to read the message and send a message but I need an email sent to them when a message is received so that nothing is missed.

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@openpbx wrote:

I have to replace a hard drive and reinstall FreePBX, don’t have much setup but I am on the trial and would like that to be added back as trunk config in the portal is grayed out until I am on a paid plan so I can’t get the settings. How do I add the trunks back? The deployment ID gets added when first setting it up but I can never get the sip station to log me in and the trunks show up as none.

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@tonysims wrote:

When I create a inbound route and point it to Sipstation DID verification I get a recorded announcement detailing my calling and called numbers which is just what I want however when I direct the inbound route to an announcement with it’s destination after playback directed to Sipstation DID it does not work.

Any thoughts? I am wondering if the calling / called information is somehow lost during the transfer to the announcement module.

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@gefflong wrote:

If this need to be posted somewhere else, let me know. I have no idea where to place it.

We have ported 4 phone numbers from Frontier to SIPStation. Each number goes to one of 4 school buildings. Each building has their own FreePBX server.

Calls from cell phones and, I believe long distance, get to our new system (FreePBX) when calling the ported numbers.

When calling any of the ported numbers from local landlines they get sent to the old system message and never get to our FreePBX server.

Any ideas on what’s going on?

I was on a call with Frontier for an hour on Friday and they were adamant that it wasn’t anything on their end.

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@markrogers54321 wrote:

Hello,

I purchased a SIPStation trunk with verified by SIPStation SMS capable DIDs and successfully manually configured the Trunks and Routes using [this Wiki page.](https://wiki.freepbx.org/display/ST/Setting+up+Standard+SIPStation+Manually+in+FreePBX). I can Imake and receive voice calls on those DIDs/Trunks. In Module Admin, I have active UCP, SIPSTATION, and SMS modules. 
I found 2 Wiki pages on setting up SMS, one under UCP and one under Modules. They say very similar things - >Admin>User Management>User X>SIPStation SMS DIDs>click DID tick box. 

On this page they also show but do not mention an “Allow SMS” toggle on the User page.
I do not see anything about SMS after checking every tab on several users in User Managment. Creating an Inbound Route with the SIPStation SMS capable DID did not change anything in that users User Management config tabs. In UCP, I don’t see SMS as an available Widget to create, but I am guessing I may have to get the User Management part figured out first.
On the 2 SMS setup Wiki pages, one mentions FPBX 14 and one mentions 12. Has SMS been deprecated on 15 ? It feels like I am missing something fundamental but I can’t see it. Thanks in advance. I’m on 15.0.17.24.

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@Bradbpw wrote:

Our inbound DIDs all accept SMS messages, the problem is some of our users don’t utilize the SMS functionality well. This can cause our customers to send a SMS and not receive an error or message that the SMS was not received or was sent to a landline. Is there a way to turn off SMS on an individual DID so that someone receives an error when sending a message to it?

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@lzcantrell wrote:

Can someone from Sangoma confirm if there is issues with Sipstation ( trunk1.freepbx.com )?

Registration has been in and out all day, 2 different customers (1 being a local machine, the other machine in the cloud)

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@GSnover wrote:

Ok - here is the scenario - I have a box with 10 Sangoma Connect users - they all want SMS, so I would need to port 10 numbers to Sipstation to have them all be SMS-Enabled all the way through?

But they are never going to be using anywhere close to that many lines for Voice - they would top out at 2-3 simultaneous at most.

So can I purchase three channels of Sipstation with 10 DID’s and SMS for all 10 DID’s?

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@macnewbold wrote:

I’m about at my wit’s end with this one. FreePBX, with a SIP Station trial currently, and everything seems to be registered fine, firewall tests say they’re passing, and I can successfully make outbound calls just fine. However, when I dial my DID at sipstation from an external phone, it acts like it’s trying to connect for a while, then the cell phone says the call failed. There’s no ringing tone at all, and no sign at all in the FreePBX logs or debugging info that the inbound call ever got to FreePBX. What’s my next step for debugging this?

Since firewall details may be related, the PBX is behind a SonicWall that is allowing ports 10000-20000 from anywhere, and 5060, 5061, and 5160 from the SIPStation trunk URLs trunk1. freepbx. com, trunk2. freepbx. com, trunktrial1. freepbx. com, trunktrial1. freepbx. com. Outbound those ports are open as well.

As far as call routing, I have a couple of extensions (102 and 103) set up and logged in (103 from my desktop with linphone, and 102 from an iOS phone with linphone app, either on the network or off network and on a VPN) and an inbound route set up to route from our DID to ext 103, and right now, I have an inbound route for anything inbound to go to ext 103 and it still doesn’t work or see any inbound call activity in the logs when I place a call.

Any suggestions on what I’m missing here?

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@defcomllc wrote:

So Ive been using Sipstation for myself and my clients for over a year now. I just use the built in Sipstation module with automatically sets up the SIP trunk and everything has worked great. With everything moving away from CHAN_SIP to PJSIP I wanted to setup my Sipstation trunk as a PJSIP trunk. Currently Sipstation is the only thing using CHANSIP on all my FreePBX and PBXact boxes.

I reached out to Sipstation about converting over they said PJSIP is supported but not through the Sipstation module. You have to set it up manually.

I cant find any wikis or previous threads about people setting up their Sipstation trunk as PJSIP.

Anyone done this before? I have never setup a SIP trunk in FreePBX or PBXact, only have used the Sipstation setup module which has been painless and working great.

Any help or directions would be appreciated on setting up Sipstation as a PJSIP trunk so I can then disable CHANSIP on all my boxes.

I currently run PJSIP on a high port in the 50K range on my FreePBX boxes.

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@defcomllc wrote:

I have been following the Bandwidth.com DDOS attack and read that SipStation uses them as their upstream carrier which is why I didnt open a ticket when issues started last week into this week.

I now see both https://status.bandwidth.com/ and https://status.sangoma.com/ says Operational and no longer says degraded so I just opened a ticket in my portal this morning.

My client is reporting incoming and outgoing calls, audio goes in and out. Can be talking for 20-30 seconds then audio drops, then it comes back after 15-20 seconds and this happens over and over throughout the call. This started last week and continued into this week in conjunction with the reported Bandwidth DDOS attack.

Anyone else experiencing this issue with SipStation trunks??

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