Happy hear you solved. Sometimes just doing something other is enough to see where is the solution.
Regards
Roberto
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Disable Intercom Auto Answer
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Upgrade from FreePBX 13 to Incredible PBX 13 with gvsip
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
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Unknown PJSIP Endpoint
I have 17 trunks and 2 extensions, all PJSIP:
Chan_PJSip Endpoints
Endpoint: <Endpoint/CID…> <State…> <Channels.>
Endpoint: 10/10 Not in use 0 of inf
Endpoint: 11/11 Not in use 0 of inf
Endpoint: alcazar Not in use 0 of inf
Endpoint: arctele Not in use 0 of inf
Endpoint: callcentric Not in use 0 of inf
Endpoint: circlenet Not in use 0 of inf
Endpoint: gvsip1 Not in use 0 of inf
Endpoint: gvsip2 Not in use 0 of inf
Endpoint: gvsip3 Not in use 0 of inf
Endpoint: ideasip Not in use 0 of inf
Endpoint: inum Not in use 0 of inf
Endpoint: ipcomms Not in use 0 of inf
Endpoint: localphone Not in use 0 of inf
Endpoint: obi110line Not in use 0 of inf
Endpoint: obi202bt Not in use 0 of inf
Endpoint: obi202btx Not in use 0 of inf
Endpoint: obi202ot Not in use 0 of inf
Endpoint: sip_broker Not in use 0 of inf
Endpoint: star Not in use 0 of inf
Objects found: 19
Reports -> Asterisk Info says there is 1 Unknown Endpoint:
PJSip Endpoints:
Available: 19
Unavailable: 0
Unknown: 1
Where can I find/identify this unknown endpoint?
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Unknown PJSIP Endpoint
Go to Asterisk Info -> Chan_PjSip Info. My system shows ‘anonymous’ and ‘dpma_endpoint’, both of which count as unknown.
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Enable disable extensions for external calls
There’s another Commercial Module that does that, more expensive, but also has more features.
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Audio issues - dropped packets
I ran a more comprehensive test using the Ring Central software. Everything looks to be good.
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Restrict internal extension calls
Any chance you could share some detail about how you got this to work by using Custom Contexts, UnknowPerson?
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Unknown PJSIP Endpoint
Thanks, Stewart, but the list I posted is Asterisk Info -> Chan_PjSip Info (I deleted everything but the first line of each entry). ‘anonymous’ is present when you have ‘Allow SIP Guests’ set to Yes and increases the Unknown count by one . I have ‘Allow SIP Guests’ set to No and don’t have an ‘anonymous’ entrypoint. I also don’t have a ‘dpma_endpoint’ (and don’t know what it is). I included everything that’s visible in my first post. I can’t account for any Unknown entrypoints.
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FreePBX 14.04.5: how to enable sip SIMPLE message
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
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Trunk Stops working randomly
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
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Sip_registrations.conf not being updated
Hi guys,
I’m using FreePBX 13.0.195.4 and am trying to change the sip host for one of my trunks…
I can make the change under TRUNKS --> SIP SETTINGS, and it is reflected there correctly, even after a reload, but no matter what I try, it will not update the /etc/asterisk/sip_registrations.conf file.
The sip_additional.conf file is updated correctly with the new information.
The file is being re-written everytime I reload freepbx, but it’s written with the old hostname, as in “register=username:secret@old-host-name”
I can’t find anywhere else in the GUI where the hostname of this trunk is used, except in the SIP SETTINGS for the trunk, and yet, it’s reflected correctly there.
I’ve even tried deleting the sip_registrations.conf file in the hope it was corrupted, but no, the same file gets recreated with the wrong information.
I’m at my wits end, so if anyone has any suggestions, they would be greatly appreciated.
Thank you,
-Michael
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HOW TO - Flowroute Trunk with Proper Use of IP Auth and new PoPs
Awesome write-up. I have been using this on a demo server for a couple of weeks with no issues. I have a PBX with ChanSIP extensions. I know it is possible to have a PJSIP trunk and ChanSIP extensions, but do you know if there are negative implications with this? (mixing ChanSIP extensions with a PJSIP Trunk)
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Hold Timer Destination
Is there a way to set a destination for a caller that is placed on hold? After 5 minutes ring the phone back instead of just dropping the call?
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Dashboard takes one minute to load
Ooooh… didn’t think about that. Good catch!
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HOW TO - Flowroute Trunk with Proper Use of IP Auth and new PoPs
Not that I know of. However, PJSIP with extensions tends to work pretty well now and is considered best practice, so I would recommend using it for both. If people do recommend against using PJSIP, it’s usually about trunks, as some sip providers don’t have great support for it. One of the side effects of using both Chan-sip and PJSIP is that you then have to have 2 ports open, probably 5060 and 5160. I like to keep ChanSIP blocked and my PJSIP port set to 5160, as that alone eliminates most of the port scanners.
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How can I play custom tone when the other end caller hangup the call?
Hi,
I would like to play a little tone to indicate the end of call when the other end hangup the phone.
Is there any easy way to achieve this?
Many thanks in advance,
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Question - Boss wants to know how many incoming calls
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
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HOW TO - Flowroute Trunk with Proper Use of IP Auth and new PoPs
I need to convert some extensions to PJSIP and see if I run into any issues. I was wondering if there was any negative side effects like increased load on the server. Would the server have to do any conversions between the 2 drivers while on a call?
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HOW TO - Flowroute Trunk with Proper Use of IP Auth and new PoPs
I don’t believe so. This is really just handling the sip packets, so there isn’t a lot of load there anyway. If you’re using 2 different codecs for the trunk and the phones, then yes, there will be a higher load while it transcodes. But codecs can be set the same for the different drivers (and I think they are by default).
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Execute AGI on AMI Originated Call
Thanks, Lorne. I did not know that from-pstn-custom was bad. I will change that and use your suggestions. I assume that will fix the AGI not running. Any suggestion on connecting the extension to the outbound call before the call is answered?
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