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PJSIP + Trunks
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Any operations with the asteriskcdrdb database lead to a fail
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Zulu merge calls
That “merge calls” issue is resolved in 14/15.0.58.7 can you test that version? thanks.
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Any chance we might see an updated blacklist module?
My evolving solution . . .
If the caller is on the 'whitelist" continue as normal, otherwise answer the phone and start recording it but don’t say anything for a few seconds while “waiting for speech”, a couple of possible scenarios. . .
A real person will say something like “hello?” or “anyone there” and then wait for you to answer, if you get silence after a short pattern of speech, then goto “2FA”
Robocallers and reverse 911 type calls start speaking and don’t stop so wait until they are done and send the recording to google STT for translation, these calls generally hangup, if they don’t then go to “2FA” . But either way, if the " STT" response is longer than n words , email/sms/push-notify the resulting STT to whatever, while possibly possitively “weighting” the STT with words like “emergency” “alert” or “fraud” and similarly negatively ‘weighting’ words like “contribute” “polical” “loan” “drug” “loan”
My “2FA” context, ask the caller a random question with an easily testable answer. like “what color is the sky?” or “what day of the week is it?” perhaps qualified by a preample of “YYou are a new caller, as such, just one time answer this question, if you answer correctly, you will never hear this again . . .” .
Send the response to STT and regex the response against “acceptable answers” using TTS. A pass adds them to my astdb whitelist “family” with date and CID,
The idea is that it is easy for a human to “pass” but harder for a machine with any DTMF (I notice some robocallers are possibly STT’ing YOUR recording like “please press 5” and doing just that)
Before anybody asks, google will roundtrip a STT/TTS of a short challenge/response in < a couple of seconds, quite acceptable to me.
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Calls drop at 30 minutes
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Zulu merge calls
Thank you for responding, Angel. We are working to define the pattern and with multiple user combinations. That’s why we did not open a ticket. Zulu is impressive to our users, which is a miracle.
Should be able to provide combination details next week. Hope that is okay.
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GS GXP2170 and End Point Manager
Hey Guys,
I have a hosted Freepbx 14 currently have all extensions are remote between two offices. I have purchased EPM to provision the phones from the server, and have setup HTTP gxp2170 template in freepbx. I can't get the GXP 2170 to read or download the config files from freepbx.
I put in the Gxp2170 upgrade and provision menu the following setup info.
upgrade server : http://myserver.com:84
upgrade username: (username from global settings)
upgrade password : (password from global settings).
Then i press provision and have tried to restart he phone to pick up the config files. I have done plenty oft Yealink phones and they are so easy to program. If anyone can guide me in the right direction with this one.
Thanks in advace!
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Group Ext is Busy
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SIP Trunk Issue
Friends,
I have been beating my head against the wall all day on this and just cannot figure it out. I am setting up a FreePBX distro with a SIP trunk from FirstComm. They claim there is no ID or secret to use, they use the external IP address for security. On my PBX, I have ETH0 as the LAN and default gateway there. ETH1 is the external IP that Firstcomm wants me to use. I have a static route for the SIP trunk IP out through ETH1. Firewall is turned off. This is the same type of config I have done for years on many different providers. I have been playing with trunk settings, adding and removing a myriad of settings and cannot get calls to move in or out. If I do a SIP debug (when I set qualify=yes), I see the PBX attempting to register and not getting a reply.
Retransmitting #1 (NAT) to 216.159.230.***:5060:
OPTIONS sip:216.159.230.*** SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK31576365;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@192.168.1.11;tag=as7834a30a
To: sip:216.159.230.***
Contact: sip:Unknown@192.168.1.11:5060
Call-ID: 4e2942ee10bc8e6b18961ced3baea28c@192.168.1.11:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-15.0.16.53(16.9.0)
Date: Fri, 05 Jun 2020 00:26:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
If I try to make a call in our out - the call just sits and never goes anywhere, just silence. The SIP provider has no clue… Im thinking that I need to put the external IP into that string its sending (instead of the ETH0 local lan .11 address), but I don’t know where to do that? Ive never had to do this before with any other provider… WHat can I send? The current trunk settings are:
type=friend
host=216.159.230.***
context=from-trunk
insecure=port,invite
trustrpid=yes
sendrpid=yes
qualify=yes
Currently inbound is blank - but I have tried many of the above settings in there too…If I ping the SIP trunk 216 address from the console, it responds fine. Traceroute shows it going out through ETH1 so I think the networking part is OK
Hopefully this makes sense - let me know what else I can give. This is the latest FreePBX distro with all updates as of 6-4-2020
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SIP Trunk Issue
Does eth1 has a public static IP? If so, you need to set it on the SIP settings on the settings menu. If the trunk is trying to register but your provider says you don’t need username or password, then no registration is needed and the trunk should not be trying to register.
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SIP Trunk Issue
eth1 is a public static IP… I will attempt that now… thanks for the quick reply.
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SIP Trunk Issue
Also if eth1 is directly connected to internet, be sure to set NAT to no on the same SIP settings page.
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SIP Trunk Issue
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SIP Trunk Issue
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Call Accounting Module Blog Post
Some questions that the wiki does not seem to cover…
Coming from a retirement village scenario, applicable to perhaps mining camps as well -
Is there an ability to, say if this is run once a month, mark the extensions to 0 (paid), keeping history? Allowing us to interrogate the database on a per user basis and charge them.
Can resident extensions view their call cost (monthly bill) history in UCP?
Is payment gateway an option that they can pay directly via UCP?
Is there an auto trunk suspension if they dont pay within X number of days? (I say trunk because they should of course have unfettered internal call usage)
(I expect some people will say go use PMS module, but theres a few problems with that argument that is not for this thread)
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SIP Trunk Issue
Are you sure that the static route is actually working? Have you tried a traceroute to the IP of the provider?
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SIP Trunk Issue
Is the default route going out eth1 or eth0?
Looks like the provider is not proxying RTP. Routing the SIP server address alone to eth1 is not enough. RTP traffic can come from/goto anywhere. If the default gateway is eth0, then all the audio traffic is going out the wrong interface.
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SIP Trunk Issue
173.161.38.17 appears to be an address from Comcast assigned to US Waterproofing in Illinois. Is any of that familiar, perhaps the location of a remote extension?
168.93.99.x is assigned to FirstComm. Are they your ISP as well as your trunking provider? If not, please explain the setup. If yes: Why do you have the default gateway on eth0? Is there a different ISP providing that?
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SIP Trunk Issue
OPTIONS sip:216.159.230.*** SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:5060
You see that via? Obviously thats not your public IP.
Also you say on chan sip settings you have set to static IP? Shouldnt it be public IP?
Better to make your default route as the public and then set the private route out the lan.
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Calls stuck in queue for over an hour?
I need a screenshot from timing and agents and capacity options.
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