How to pass calls to users #23
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For freepbx17 you need to add "trunk-dial-with-exten" and "dialOne-with-exten" contexts in the config like that: dialplan:
incoming_context:
- from-pstn
outgoing_context:
- macro-dialout-trunk
- trunk-dial-with-exten
ext_context:
- macro-dial-one
- macro-dial
- dialOne-with-exten
dial_context: from-internal
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Unfortunately, it still isn't passing calls to SuiteCRM, at least that I can see. Maybe I need something else installed on SuiteCRM, the only call module I have is from this one, otherwise the SuiteCRM install is completely fresh. This is what I see when running Asterlink from CLI during a call, not sure if it helps. ./asterlink INFO[0000] AsterLink |
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What is the trunk context on freepbx ? It must be |
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from-pstn-toheader I have already tried changing the incoming_context in conf.yml as well, to no avail, same behavior, same thing in Asterlink CLI. It gets the incoming phone number and DID, but no extension numbers. It does get the incoming phone number. It also does not recognize when the call is passed from the IVR, to an extension. It only recognizes when the call hits the PBX and when the call is disconnected, the callers phone number and the trunk's phone number. Is it maybe possible to use the trunk phone number, instead of the extension number? |
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I guess your trunk provider places calls to "s" extension and so that why you need to use The problem is the "s" extension is filtered here: Line 197 in 302e7f9 You can try to "fix" it by hardcoding DID value in the code like that: // incoming call
func (a *amiConnector) onNewchannel(e map[string]string) {
+ if a.isContext(e["Context"], a.cfg.DP.In) && e["Exten"] == "s" {
+ e["Exten"] = "1234567MYDID"
+ }
if /*_, ok := IncomingContext[e["Context"]]; !ok || */ e["Exten"] == "s" { // TODO: review this 's'!
return
}You can also specify both dialplan:
incoming_context:
- from-pstn
- from-pstn-toheader
outgoing_context:
- macro-dialout-trunk
- trunk-dial-with-exten
ext_context:
- macro-dial-one
- macro-dial
- dialOne-with-exten
dial_context: from-internalIs that trunk with registration ? If so, can you check if changing trunk's "Contact user" to your DID will ask the trunk provider to send calls to that extension and not "s" ? |
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Changing trunk's "Contact user" to DID made the call pop up in the log with timestamp and phone number in "Calls" section! No need to edit ami_connector.go. Is it possible to have calls show up in the browser or to be able to click to call back? ./asterlink |
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Check README_suitecrm.md. |
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It is configured according to the example suitecrm: http://127.0.0.1:5678 is address given in gui and endpoint token is same in both gui and conf.yml |
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Is asterlink and suitecrm on the same server ? If so, enable proxy mode in gui. Address given in gui will be used by both suitecrm and users browsers to connect to asterlink, so it must be accessible for them. With current setup: Without proxy mode enabled: With proxy mode enabled:
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proxy mode is enabled and I think working because calls are logging ext= only shows up in Asterlink CLI after "Dial stop" |
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Is your extension in I will need AMI dump to debug this issue. AMI dump can contain some private info, like CallerID and DID numbers
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I emailed you it. Big file! |
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For ring groups you need to also add |
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Okay that made it much better. It now logs calls as soon as they come in, in the "Calls" GUI of SuiteCRM, not after they are hung up, and they are also assigned to the correct user. Asterlink also now reports "Answer" when the physical phone is answered, and the duration of the call is recorded in the "Calls" GUI of SuiteCRM. It also recognizes the extension both as soon as the user exits the IVR and as soon as the call is hung up, it sees the extension number both times. With "- from-pstn-toheader" in the incoming_context, it does not seem to record the extension number, even if "- from-pstn" is also there. But it records the extension number fine with only "- from-pstn" along with the number in the Contact User in FreePBX GUI. However, there is still no click to call or popup when the call comes in. I'm wondering if I'm missing something, a module or something for this. I have the endpoint URL configured as well as the proxy enabled. |
If your endpoint url settings is correct, you should see " Are both suitecrm and asterlink on the same server machine ? You can access Also, if you click on the contact phone in suitecrm, it shows you "Unable to dial" popup ?
If contact user works, you can set |
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Yes, they are running on the same machine. There is no response from https://SuiteCRM/index.php?entryPoint=AsterLink&action=stream unless I set the url to localhost or 127.0.0.1 or the machines remote IP, at which point it says "Extension not found for user id". Asterlink also responds with "ERRO[0000] invalid character '<' looking for beginning of value suite=true" when starting this way even though it does continue to run. Should I try disabling HTTPS? Is that the issue? |
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HTTPS shouldn't be an issue. You can check if your suitecrm has connection with asterlink by running tcpdump on its server. It should look like this: Also, suitecrm webserver needs php-curl module installed. |
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is there any way to use asterlink 0.5.1 with suitecrm version 8.7? i tried to install but it says copy error |
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I don't think so - I had to delete SuiteCRM and reinstall as 8.5: at which point the module loaded right up in module loader, no copy() error |
You should've created a new issue. Well, whatever. 8.7 has a bug where Here is the archive with workaround: Asterlink 0.5.1 still works with 8.7.1. |
sorry , i will create issues next time , i will try with fix module , thanks a lot |
SuiteCRM 8.5
The asterisk is running on freepbx 17
Asterlink can see the calls... I think the issue is in the SuiteCRM Asterisk Extension, the Asterisk Extension in user profile not being picked up by the FreePBX extension. Asterlink CLI shows the user extension being loaded from SuiteCRM as 987:1 (given extension number in suitecrm gui is 987, asterlink is adding the :1)
Let me know if any logs would help please
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