Is it possible to configure FreePBX so that when a caller dials an extension, the name appears on the display during the call (while it is ringing), instead of just showing the extension number? The person receiving the call can see the name clearly, but the caller only sees the extension. When the phones were on the OpenStage system, this feature worked. What should I do, and is this feasible?
In the extension options, the “Send RPID to Send P-Asserted-Identity header” is already configured.
I think this depends on the phone you are using to dial from. Not sure that we’ve had that working in any of our instances. We only see the number that’s being dialed. Maybe somebody here would have a better answer if they have experience with a device that matches a dialed number to a phone book entry and then shows the information from that entry.
When making a call, the caller sees the name instead of the extension, but only when the other party picks up the phone. This is probably the default behavior, and that’s fine. To display the name while calling, I probably need to connect the phone to the directory on the server (LDAP). I haven’t done that yet, so I’ll need to look into it.
Hvala Igore
This setup ensures that the CONNECTEDLINE information is set globally, allowing the caller to see the name of the person they are calling before the ringing phase begins. The macro retrieves the caller’s name from the database and applies it as CONNECTEDLINE(name) before initiating the dial process.
This works for me. Is this configuration correct, or would anyone suggest any changes?
Everything seems to be working fine for now, but is there anything I might have missed that could cause issues later?
I’m not sure why you’re doing this as CONNECTEDLINE is done already in the existing dialplan. Additionally both chan_sip and app_macro are dead in Asterisk moving forward. I’d change your extensions to chan_pjsip and gosub’s.
Also, your addition of [from-internal] in the _custom.conf file is not the proper way to do this and will most like break down the road.
If I don’t set the CONNECTEDLINE before the ringing phase, it doesn’t display the name of the person I’m calling before the call is answered. This way, it works for me.
Strongly recommend against using dialplan extensions like _4XXX in from-internal-custom. Every line you put there, supersedes an autogenerated line by freepbx, there will be conflicts that may not be immediately evident.
There are dialplan hooks provided for this exact purpose, if you need them.
How about you provide us with a debug of a call that is just using the standard dialplan. This is already something handled in standard dialplan. So we should see what is happening and find the problem.
We should have a verbose call output from Asterisk CLI and you should also get a debug from chan_sip.
Thank you all, I will send it as soon as I am able. This may work normally with other phones, but it won’t work with OpenStage without this configuration.
In file “with_standard_dialplan”
[2024-09-12 18:43:58] VERBOSE[27563][C-00001293] app_dial.c: Connected line update to PJSIP/400-00002a59 prevented.
Why is this?
For some reason, I found a fork here, but I assume the standard code is the same.
I haven’t checked if FreePBX sets this, or if the dialplan overrides it.
In the former case, I’m sure you can override it.
[2024-09-12 18:43:58] VERBOSE[27563][C-00001293] pbx.c: Executing [s@macro-dial:30] Dial("PJSIP/400-00002a59", "Local/FMPR-456@from-internal&Local/FMGL-702#@from-internal,22,HhTtrIM(auto-blkvm)b(func-apply-sipheaders^s^1),") in new stack
Well I can see the major difference here. The extension you are dialing, 456 has Follow Me enabled which is then calling 456 and 702. Even more 702 is entered as 702# which means, also use 702’s Follow Me list / call forwarding.
What this means is, your gosub kludge is ignoring all the extension’s settings (including Follow Me) and just directly dialling the PJSIP endpoint for the extension. However, when using the standard dialplan which respects all of the extension settings which triggers Follow Me it is now dialling two destinations via Local channels. That is going to make a difference.
gosub kludge: PJSIP Channel A is calling PJSIP Channel B…Channel B is passing CONNECTEDLINE back to Channel A.
Standard Dialplan: PJSIP Channel A (400) is spawning two pairs of Local channels which in the case of 456 on its own looks like
PJSIP Channel A (400) --> Local Channel 1, Local Channel 1 --> Local Channel 2, Local Channel 2 --> PJSIP Channel B (456)
PJSIP Channel A (400) --> Local Channel 1, Local Channel 1 --> Local Channel 2, Local Channel 2 --> PJSIP Channel C (702)
It’s very possible the local channels are preventing the CONNECTEDLINE update being sent back to the original caller.
That is why I also requested a sip debug so we could see if the SIP replies/updates are being sent back to the calling endpoint. You can generate a pcap directly with pjsip set logger pcap /home/asterisk/call.pcap
You can also test this by turning off Follow Me which will remove the use of local channels and see if the CONNECTEDLINE details update the calling endpoint.
Thank you very much. I had already tested without the Follow Me option and without any additional configuration (using the standard dialplan), and it didn’t work. I see that with this configuration using gosub, Follow Me won’t work… The only configuration that works for me is:
