Frustrated: Cant get Early Media to work!

NoFate · October 25, 2021, 3:23pm

ok, gonna test it later
can you give me a last hand, how i can use early video, if i call a group, that video is present on all sip phones?
thnx

NoFate · October 25, 2021, 5:02pm

@david551 , was testing the dialplan, but what happens , so if i call 2002 from 2000 , i see early video , thats good… but when i end the call (before i pickup) , 2000 starts calling again (without early video)…

what could be wrong?

[from-internal-custom]
exten => 2002,1,Progress()
same => n,Dial(SIP/2002)

david551 · October 25, 2021, 5:20pm

That’s a problem with the device.

Or you haven’t explained it clearly, in which case the sip set debug on logs are needed to see what is really happening.

NoFate · October 25, 2021, 5:23pm

ok, will enable debug, and post results later

any idea how i can make it happen on ringgroups? do you think its possible?
or another approach is to setup max contacts for pjsip extensions, and register all my softphones with same extensions… will that work?

david551 · October 25, 2021, 5:38pm

Asterisk has no such thing as a ring group.

As I already said the code that does early media bridging for the Queue application is different from that for the Dial application.

It looks like I found an old GIT repository before, and the current one does indicate that DIal supports video early media. However, it also indicates that early media is suppressed, in both directions, when there is more than one outgoing channel.

Queue() doesn’t seem to support early media, at all.

I’d expect Dial to bridge in the direction you want, when there re multiple callees, but not in the reverse direction.

EkFudrek · October 25, 2021, 6:24pm

linphone has small technical paper about what a server has to do what you want. It looks as if video intercoms have to explicitly supported. It’s more than a simple video call, but also not a video conference:

NoFate · October 25, 2021, 6:30pm

hi @david551

thats the log, with this code :
[from-internal-custom]
exten => 2002,1,Progress()
same => n,Dial(SIP/2002)

created 2 extensions
i call from 2001 to 2002 , on the 2002 i reject the call , then it starts ringing again

if i remove that code, if i then reject the call , there is no second call

david551 · October 25, 2021, 7:18pm

There is no evidence of a follow-on call in the log.

NoFate · October 25, 2021, 7:25pm

you see here :

[2021-10-25 17:56:33] VERBOSE[8266][C-00000003] app_dial.c: Called SIP/2002
then i decline:
[2021-10-25 17:56:38] VERBOSE[2221][C-00000003] chan_sip.c: Got SIP response 603 “Decline” back from 10.8.0.3:4204
and then it calls again:
[2021-10-25 17:56:38] VERBOSE[8266][C-00000003] app_dial.c: Called SIP/2002

david551 · October 25, 2021, 7:34pm

See it now. That is FreePBX’s doing. It looks as though it is trying to send the call to voicemail, but, maybe, that has proved impossible and it is trying to re-present the call, to rescue it. In any case, I’m not familiar with the FreePBX logic on such things and AGAIN, this is the wrong forum for FreePBX related questions.

NoFate · October 25, 2021, 8:25pm

ok, gonna setup this week an asteriks server and test again…
going to make also a dialgroup

https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Function_DIALGROUP

will a dialgroup call all extensions at the same time in that group? wll early media work then?
can you help me how i need to change that dialplan then?

thnx, really appreciated

david551 · October 25, 2021, 8:31pm

DIALGROUP is just a special purpose string manipulation operator for strings with & between elements. Dial will dial endpoints separated by & in parallel.

NoFate · October 25, 2021, 8:33pm

so not possible then to have early media on multipe endpoints at same time

david551 · October 25, 2021, 8:41pm

github.com

asterisk/asterisk/blob/master/apps/app_dial.c#L1222

    
      
          	const int ignore_cc,
          	struct ast_party_id *forced_clid, struct ast_party_id *stored_clid,
          	struct ast_bridge_config *config)
          {
          	struct cause_args num = *num_in;
          	int prestart = num.busy + num.congestion + num.nochan;
          	int orig = *to;
          	struct ast_channel *peer = NULL;
          	struct chanlist *outgoing = AST_LIST_FIRST(out_chans);
          	/* single is set if only one destination is enabled */
          	int single = outgoing && !AST_LIST_NEXT(outgoing, node);
          	int caller_entertained = outgoing
          		&& ast_test_flag64(outgoing, OPT_MUSICBACK | OPT_RINGBACK);
          	struct ast_str *featurecode = ast_str_alloca(AST_FEATURE_MAX_LEN + 1);
          	int cc_recall_core_id;
          	int is_cc_recall;
          	int cc_frame_received = 0;
          	int num_ringing = 0;
          	int sent_ring = 0;
          	int sent_progress = 0;
          	struct timeval start = ast_tvnow();

github.com

asterisk/asterisk/blob/master/apps/app_dial.c#L1791

    
      
          				case AST_FRAME_HTML:
          					/* Forward HTML stuff */
          					if (!ast_test_flag64(o, DIAL_NOFORWARDHTML)
          						&& ast_channel_sendhtml(o->chan, f->subclass.integer, f->data.ptr, f->datalen) == -1) {
          						ast_log(LOG_WARNING, "Unable to send URL\n");
          					}
          					break;
          				case AST_FRAME_VIDEO:
          				case AST_FRAME_VOICE:
          				case AST_FRAME_IMAGE:
          					if (!single || caller_entertained) {
          						/*
          						 * We are calling multiple parties or caller is being
          						 * entertained and has thus not been made compatible.
          						 * No need to check any other called parties.
          						 */
          						goto skip_frame;
          					}
          					/* Fall through */
          				case AST_FRAME_TEXT:
          				case AST_FRAME_DTMF_BEGIN:

NoFate · October 25, 2021, 8:45pm

thats chinese for me

milos_ladni · October 25, 2021, 9:00pm

Hi @NoFate ,

the first issue with recalling:
I think you miss ‘same => n,Hangup’ at the end of your dialplan.

The second issue (Early Media in group call):
I am also looking for a solution where we can have video in early media.
Does anyone have some idea how to get it working?

NoFate · October 25, 2021, 9:04pm

hey @milos_ladni , that fixed it!!

[from-internal-custom]
exten => 2002,1,Progress()
same => n,Dial(SIP/2002)
same => n,Hangup

now, only left to have early media in a group/multiple softphones at once
should be possible no? isnt it possible to change that diaplan to call multiple extensions at same time?

david551 · October 25, 2021, 10:02pm

I hadn’t considered that FreePBX might inlcude the custom context in a way that meant the standard code was still live. What was happening is that you were continuing with the standard FreePBX code from priority 3. If you’d used the code on bare Asterisk it would have worked, because there would have been no priority 3.

I’m afraid I assumed you’d done something to get the Progress into the FreePBX logic, rather and that was why your log was full of FreePBX code. Otherwise I would have spotted this earlier.

NoFate · October 26, 2021, 6:17am

np
btw, if early video is not suppported in dialgroup/ringgroup…
is there maybe a way to do it manually, by injecting an RTSP stream url? or atleast somekind of still-image? adding like call-info into header?

allthough i dont know the syntac how to inject it to that dialplan

milos_ladni · October 26, 2021, 6:32am

I am also thinking like you

Injecting a custom header in sip invite SDP with some RTSP or HTTP URL is not a problem.
I have already done something similar for another purpose.
If you develop your own sip devices you can implement this easily.
And this should work between your devices.
But the problem is how to get it working with a standard third-party sip phone.

Still-image is half solution but unfortunately, i am afraid it is also not possible in standard sip.