Depends on your firewall/router. Look for a section entitled port forwarding
Are the phones on the same local network as the PBX?
No. Phones in 10.69.0.0/24
Asterisk in 10.69.10.0/24
So they are local as in not going over the Internet. If you cant make extension to extension calls with audio, you dont have things configured right on the pbx.
Describe how that is setup.
I can do calls with audio. Extension to extension. Extension to real number. Real number to my telephones.
But do all your calls drop after 31 seconds?
I did a quick Google search and found these similar, but resolved issues. Did you have a chance to look at these:
I have a FreePBX version 14, the calls from outside to inside fall in 32 seconds, regardless if there is voice or not, I researched it, checked all possibilities plus nothing at all, my FreePBX NAT identifies the external IP normally, the RTP ports 10000-20000, SIP 5060 are open, I put in sip.conf the option canreinvite = no, in advanced settings the NAT = no or yes, I changed the touch times, and nothing changed, in SIP settings the timeout RTP = 30s Hold Timeout = 300 Keep Alive = 0, I made th…
Hello everyone. I’ve seen the “calls dropped after 30 seconds” problem all over the place on older posts in this forum and on other forums using asterisk-based PBXs. I’m experiencing the same thing. It looks like the issue is firewall based or NAT based in most cases. I tried port forwarding port 5060 on my FreePBX local IP address through my router as well. I also tried an inbound call with both the PBX and router firewalls disabled and the call still cut off after 30 seconds.
Nothing seems to…
Outbound calls drop after 30 sec. i’m at a loss. I see there are other threads on this problem, and I have tried all the suggested solns. If someone with more experience could look at the log and tell me what is going on i would be greatful.
[2018-08-22 20:41:00] VERBOSE[C-00000016] app_dial.c: Called SIP/Flowrout_Trunk/16203883279
[2018-08-22 20:41:03] VERBOSE[C-00000016] app_dial.c: SIP/Flowrout_Trunk-0000000f is ringing
[2018-08-22 20:41:03] VERBOSE[C-00000016] …
I am new to freepbx from an Trixbox background.
I recently got into a situation where I have needed to reconfigure my switch from scratch (HD failure in the previous one) and found that Trixbox CE is no longer available and so I switched to freepbx. The network configuration is the same WRT internal NATted IP addresses, DID providers, port forwarding etc. Prior to the drive failure this worked well, but since the upgrade, I have all incoming calls dropped after about 30 seconds or so.
I think …
Can anyone help with this, I have installed the new system over the weekend and now calls are cutting off after 30 seconds. I have checked the logs and it appears that my system is hanging up.
All are outbound calls.
There have been about 300 outbound calls today and about 20 of them have failed with this issue.
[Dec 6 13:55:41] VERBOSE pbx.c: – Executing [
[email protected]:1] Macro(“SIP/6002-000000bd”, “hangupcall,”) in new stack
[Dec 6 13:55:41] VERBOSE pbx.c…
Had The Same Problem
Go to (Setting>Asterisk Sip Setting) check your “Local Networks” I leave this blank, It interfere with my trunks Go to (> Chan PJSip Settings) on the bottom you’ll fined "External IP Address " and “Local network” set your nat setting here. Work for me
@comtech That is a shotgun approach to this. Did you read any of those to determine if they were duplicate or even relevant to this issue at all?
@necto_random Alright so you probably have your PBX configured improperly for the network setup. If all the calls, including those on the local network, are dropping after 32 seconds the RTP/media IPs details are either not correct or something in the local network is messing with the traffic.
What do you have in your Asterisk SIP Settings? All three tabs: General, Chan_SIP, Chan_PJSIP
Now server is down) I’m at home.
I write about this at the mondey.
When i call all lines is busy.
Call is in freepbx queue, but it drop at 32 second too.
Now only incoming calls falls. Server freepbx was only rebooted.
Why don’t you follow the instructions or answer our questions here?
i do it, i add advanced in pjsip, added local interfaces and ports fnd it wark correctly
You’re still not answering our questions about your setup and what is actually configured. You’re having network/NAT issues and to be honest, you’re using a Mikrotik so that means the only way this is screwed up is because the Mikrotik has been setup wrong. Those things Just Work ™ with SIP setups and I’ve never seen them have issues like this unless they were messed with and rules are screwed up.
You need to actually answer our questions and provide us with the details we asked for.
Вы все еще не отвечаете на наши вопросы о вашей настройке и о том, что на самом деле настроено. У вас проблемы с сетью / NAT, и, честно говоря, вы используете Mikrotik, поэтому единственный способ, которым это облажалось, - это неправильная настройка Mikrotik. Эти вещи просто работают с настройками SIP, и я никогда не видел, чтобы у них возникали подобные проблемы, если они не были испорчены, а правила облажаны.
Вы должны ответить на наши вопросы и предоставить нам детали, которые мы просили.
now all works correctly. i solv problem
I addad ports and networks for pjsip
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