If you get your Grandstream registered, you should be able to make inbound/outbound calls. You would just need to create the extensions and make sure the trunk connected to an outbound route that has a routable dial pattern. This can be done in the GUI.
A quick Google search revealed:
https://wiki.freepbx.org/display/FOP/Configuring+a+Grandstream+GXW-410X+Device+to+act+as+an+FXO+Gateway
These cover registering each jack as a SIP account or setting up the Grandstream as a PEER gateway.
To confirm, if the number were 03000, you are only seeing 03000? Not 03000-1 and 03000-2?
Bandwidth is good (100MbitD/20MbitU) at the customer site and really good (1GbitU/D) at my CoLo - I am just trying to find out if anybody has had any real world problems because it seems to work perfectly.
Couple of things I noticed while troubleshooting:
I had knocked the Qualify time down to 20sec - this seemed to be necessary to get 95% of the phones to have a stable connection - it was in the end only 3 phones that I could never get stable - switching to TCP fixed that and in fact I switched back to a 60sec qualify and they were all still solid.
Voice Quality seems to have improved - without prompting, I got two comments that they sounded better - I wasn’t even asking.
The customer connection is Comcast, and they are terrible here as far as reliability and consistency goes - so maybe it was necessary in this case because of the circuit, but whatever - this is going to be my go-to solution in the future when this is a problem.
Nevermind I was just thinking EPM worked a little differently than it did. I was thinking the custom template for that extension would apply to both but I had to custom edit both the -1 and -2
That is my conclusion also - they seem to be a good provider, but for Hosted sites, the only phones they completely support (everything works) is Polycom - not my favorite phones, but whatever.
Hi everyone. We are running FreePBX 13.0.194.2 and using pjsip for our trunks. Even though we have max retires set to 0 on the trunks, if we have an issue that prevents registration (an Internet outage for example) the trunks stop attempting to register after a certain number of attempts. When the Internet comes back, the trunks do not try to re-register. Am I missing something here? Is there another setting that needs to be set?
2018-04-16 02:30:15] WARNING[15782] res_pjsip_outbound_registration.c: Maximum retries reached when attempting outbound registration to ‘sip:username@sipserver:5060’ with client ‘sip:username@server:5060’, stopping registration attempt
Thanks Stewart1. The issue was resolved by increasing the RTP Hold Timeout to 1800 seconds. However I am going to try to do a bit more digging because the dropped calls do not happen with any of the Yealink handsets but rather only the Cisco 7942’s.
When I look at the packet capture between a Cisco 7942 and the PBX there is an OPTIONS packet being sent every minutes from FreePBX server to handset and the handset is returning a 200 OK but after the 5th OPTIONS and 200 the PBX is hanging up the call by sending a BYE. I do not see the same BYE with the Yealinks.
Thanks again for the assist!
Yes the time is right Appointment reminder status switch from running to not running when the time come but still calling out
Current PBX Version 14.0.2.14
PBX Firmware: 12.7.4-1804-1.sng7
PBX Service Pack: 1.0.0.0
I’m looking for a way to extend the time a user has to enter an extension to transfer a call.
I have a user who is a backup for the main receptionist and received overflow calls. She has had to transfer a couple of call this morning to voicemail using *123 for example and the first time she had to lookup the desired extension but by the time she found it and had already pressed * it errored out. If she dials it all in one sweep it works fine. This was an employee she didn’t have a BLF for.
Is there a way to give her or other users a little longer to enter extensions to transfer before it will error?
i have 5 inbound lines that connected to GXW without ability to make outbound calls. One line that have this ability. All lines are registered with sip accounts on freepbx. GXW uses unconventional call forward for incoming calls.
All instructions for freepbx that i found talks about "outbound routes " to work trough trunk. But my GXW doesn’t connect it. So i want to enable outbound call in some other way.
Hellow everyone,
I am new to this forum. I am looking for a solution. I have asterisk 13 and freepbx 13 running on Centos 7. Fanvil phones as sip clients. When we have missed calles on phones, and operator redials the call I whant to replace first 4 digits with one digit. Can anyone suggest any solution?
Thank you
If they are all registered (OK) SIP extensions, you should be able to use the trunks in FreePBX. You would need to have a working trunk in FreePBX. That trunk needs to be part of an outbound route in FreePBX. The outbound route would need a dial pattern that matches what you are attempting to dial from one of the Grandstream lines.
https://wiki.freepbx.org/display/FPG/Outbound+Routes+Module+User+Guide
https://wiki.freepbx.org/display/FPG/Trunks+Module
Has anyone figured out a way to get a call notifying a user of a pending voicemail and allow them to retrieve it or have the Voicemail forwarded to a cell phone.
How sure are you that setting max_retries to 0 means unlimited? I can’t find any docs to support this, you may be better off with a very high number. The tool tip states a max value of 1000000, but I can’t find any docs to support that either.
Dialed Number Manipulation Rules in the trunk settings?
Here it is – it is in the phone
Live Dialpad*
The “Live Dialpad” option on the IP phone turns the Live Dial Pad mode ON or OFF. With live dial pad ON, the 6731i IP
phone automatically dials out and turns ON hands free mode as soon as a dial pad key or programmable key is pressed.
With live dial pad OFF, if you dial a number while the phone is on-hook, lifting the receiver or pressing the initiates
a call to that number.
*Availability of feature dependant on your phone system or service provider.
You can enable/disable the live dialpad using the IP Phone UI only.
Enabling/Disabling Live Dialpad
Press on the phone to enter the Options List.
Select Preferences.
Select Live Dialpad.
Use the Change key to turn the live dialpad ON or OFF.
Press Done to save your selection.
So, problem solved.
There is a paid commercial module that does this:
I cant recall where I came across that information, possibly on this forum somewhere. If this is not correct, I will try setting the max retries to a very high number to see if that works. I am being told on another forum, that this is a known issue with pjsip and I should try using chan-sip.
We are running FreePBX 13.0.194.5
NAT = Yes, Static IP
We are experiencing lots of very high active calls anywhere from over a hundred to several hundred. This is displayed from the dashboard when looking at the statistics.
When looking at the full log file I see lots of the following:
[2018-04-16 08:41:26] DEBUG[35033][C-0000033e] format_wav.c: Skipping unknown block ‘LIST’
[2018-04-16 08:37:35] WARNING[34090][C-0000031e] chan_sip.c: This function can only be used on SIP channels.
[2018-04-16 08:37:35] ERROR[34030][C-000002ef] res_pjsip_header_funcs.c: This function requires a PJSIP channel.
[2018-04-16 08:37:35] WARNING[34092][C-0000031e] app_dial.c: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
NOTICE[13090] chan_sip.c: Disconnecting call ‘SIP/TwilioIn4-00000705’ for lack of RTP activity in 31 seconds
[2018-04-16 08:37:49] WARNING[13090] chan_sip.c: Autodestruct on dialog ‘3dfce109341112224e7a60c8525d0e95@10.0.1.26:5060’ with owner SIP/1322-00000706 in place (Method: BYE). Rescheduling destruction for 10000 ms
Other symptoms we are seeing is when you place a call from the outside to the PBX, I hear the bleep and then a long period of silence and then eventually I hear the IVR announcement play.
When transferring a call, there is a long delay in the pickup.
This is a system that has been running very stable for quite some time. Two things have recently changed. We recently installed the datadog-agent v6 but we have since removed that agent. The other is that we moved the cdr and cel tables to another folder with more disk space. I had initially created a symlink to the whole directory but that apparently according to some docs I read was not a good thing to do. I have since put the .FMT files back to where they were and only symlinked the MYI and MYD files and initially that seemed to make everything run fine. After a couple days though, we are experiencing the issue again. I have checked the tables by running “check table cdr” and “check table cel” and they show as OK.
Any suggestions?
