When they did away with the official “Community Leader” designator, they established a “Leader” tag that they could give to amateurs so that they could communicate with them, and they would in turn communicate with the group. The leader tag gives one slightly more access to Sangoma resources than the average person. Slightly.
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Serious Development Question
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Internal call answered externaly
Did anyone but me notice the “PJSIP/anonymous” part? That’s almost always not good.
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Reset feature codes for extension to default
The Rule of Least Astonishment would lead me to believe that a Factory Reset from EPM should clear all of this. EPM will reload the phone with the configuration your EPM designates, and away you go.
I don’t use Sangoma phones, so this is a real question - isn’t there a way to lock the phone configuration so that only an admin can reset them.
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Make call and play sound from external system
So if you must choose would you go for AMI or ARI to do this?
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Upgrade Freepbx 13 to 14 Technical limitations prevent 6 to 7
This topic was automatically closed 365 days after the last reply. New replies are no longer allowed.
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Dial Pattern Problems
This not correct. Outbound routes are tried in order from top to bottom which is why the GUI allows you to sort them and why it preserves the order that you set. When making an outbound call, the dial string is compared to all the dial patterns in the topmost outbound route. If there is a match, no other outbound routes will be tried. If there is no match (and only if there is no match), it will go on to the next outbound route in order with the same check. If the dial pattern matches on an outbound route at the top, it makes no difference what dial patterns are specified in the routes that follow, they will be ignored.
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Transfer to VM REST-API does not accept direct extension entry
No, my issue ticket was put in and no real priority to get this corrected. The deficiency was acknowledged though. Many others have come up with various workarounds to get an easy “Transfer to VM”.
I came up with a workaround that is easy. We are using Sangoma phones, and the Endpoint Manager.
We are using the short & long BLF press. In a rather small office, we will setup each extension as a BLF on the phone. A short press will do a transfer (either blind or attended, depending on your other FreePBX settings), a long press will do a Transfer to VM (*+ext).
To set the long press behavior do the following: From FreePBX navigate to Settings -> Endpoint Manager -> Brands, Sangoma -> Options Tab. In the Options look for “BLF Long Press Idle State” (this is the state of the associated BLF extension). Put an “*” in this field. Click Apply. Be sure to then Save & Rebuild Configurations in Extension Mapping. Send Config to extension or allow it to pick it up on its own.
I hope that this helps. If you are still looking for a solution not using a BLF key, then there are several forum threads that suggest programming a phone key as a DTMF key, with the appropriate Transfer to VM activation with a “+” following, etc.
I also discussed both solutions, gathered from others in this thread:
Good luck!
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FOP with ZULU softphone - not great experience so far
Can you define what supervisor functionality is to you? Is it just the ability to see presence? I haven’t used FOP in a while, but can you add the 90-version of the extension in FOP?
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Transfer to VM REST-API does not accept direct extension entry
When you press the Transfer to VM button, the first thing you must do is press the ‘change’ button at the bottom. This will allow you to enter a full extension number or a partial number to get a list of matches to choose from.
I need to update the wiki to reflect this.
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Callers ID does not show phone number
They will just reroute me to a branch in my country then have me create a ticket then I will have to wait 3-5 working days. That’s how it works here unfortunately.
But that aside I went into my chan_dahdi.conf and strangely there are no parameters defined for handling the CID, there was only 1
“usercallerid=yes”
I want to edit it but I’m still clueless as to what cidsignalling method my provider uses.
Would editing the file cause my calls to drop (I don’t think it would since it only relates to the caller ID but I need confirmation). If it doesn’t then I will try to edit for all 4 methods to see which one works for me.
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Callers ID does not show phone number
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Callers ID does not show phone number
lol Yes I came across this page this morning. I just want to know if editing the config file would some how cause some down time.
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FOP with ZULU softphone - not great experience so far
I need to do the recent update but nothing so far in the changelogs or FOP updates have made FOP2 work with PJSIP yet. That is to say, FOP2 works with PJSIP but it has not logic in it like it does for the older drivers (Chan_SIP, IAX) to look for the endpoint configs to auto create buttons, etc in FOP2.
Zulu softphone extension’s that have the 90 prefix are PJSIP extensions. FOP2 isn’t going to create buttons for them automatically or the buttons won’t be created with the proper tech if they are.
I have to use FOP2 without using the FreePBX “tie-in” so I can create buttons myself for PJSIP endpoints.Transfer, Hold, Park, etc all work in my testing that way.
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Callers ID does not show phone number
It technically wont because nothing will change as far as asterisk is concerned .
In reality though, it will, because every time you commit a change to the Dahdi driver, then the whole of asterisk needs to be stopped and started, not just reloaded for it to take effect
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Transfer to VM REST-API does not accept direct extension entry
I am using the same workaround-- it’s just weird that they’d provide a feature that essentially doesn’t work. I would never show a non-techie end user that app.
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Make call and play sound from external system
I’d probably go AMI, but I’m old-school. You’re not going to go wrong either way.
Look around for code APIs for whatever language you are using and see which of them you want to use. There are several language support packages for either interface. Some are a little easier to use than others, so look for one that makes it easy for you in the language you are using.
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IVR Key Press / Digit Timeout can't be changed from 3 seconds
I’m trying to increase the digit timeout to 10 seconds. I tried enabling force strict dial timeout then set the Timeout drop down value to 10. IVR digit timeout stays 3.000 seconds.
[2019-08-29 07:47:57] VERBOSE[2916][C-00076876] pbx.c: Executing [s@ivr-57:10] Set(“SIP/XXXXXXXXX-000236dd”, “TIMEOUT(digit)=3”) in new stack
[2019-08-29 07:47:57] VERBOSE[2916][C-00076876] func_timeout.c: Digit timeout set to 3.000
I believe this is set by the WaitExten() application, which I can’t seem to modify via GUI. Can this value be changed?
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Serious Development Question
cynjut Dave Burgess Leader
Could you elaborate “slightly more” on what this leader tag is about?
Who else has it?
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Serious Development Question
https://community.freepbx.org/badges/4/leader?username=cynjut
There are 49 other people that have it.
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Dial Pattern Problems
Thanks yes this is exactly the case, only if we enter a catchall at the match pattern, we are able to get outbound calls going using only the top outbound route. I will post some screenshots later on.
This is the only way we can get it to work so far, using a catchall, the last 2XX are for the internal extensions to be able to call using there own phone number. instead of calling other people randomly, this also seems to be working. for example if my phone number is 044565424 what will i need to put in for it to only use this outbound route.
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