Remotely bridging without RTP

Asterisk Asterisk Support
1

Hello all,

my asterisk does twno different calls (with SIP) towards some PSTN phones (therefore not VoiP) and after it has leave a message for each calls, it does remotely bridging correctly, but RTP media goes through asterisk.
I would like to do go rtp directly between phones?

I have read that I have to use directmedia=yes.

Unfortunately I still have the problem.
Can someone help me please?

Thanks in advance
Jonh

2

Disable recording or dial options like tTrXxwW

3

Hi Navaismo,
thanks for reply.
I have no one of the dial options above. I try to explain what I do.
From AMI interface I place a call, asterisk leave a message and then make a call to another phone, then leave another message and after place phones in remotely bridging.
In this mode the RTP media goes through asterisk.
maybe the problem is the call from AMI?

Ami command:
Action: Originate
Channel: SIP/$channeltocall/$extension
Exten: s
Context: test
Priority: 1
Async: yes

Thanks
Jonh

4

And how it looks the Test context?

5

Set the right codec on the Originate. By default Originate is done with SLIN, which normally forces a transcoding, which is another reason for direct media being inhibited.

6

Here it is…

exten => s,1,answer
same => 2,Wait(1)
same => n,Set(CALLERID(name)=$channeltocall)
same => n,Dial(SIP/$channeltocall/$extension)
same => n,Hangup()

Thanks

7

Hi david55

as you suggested, I setted codecs correctly, but they do not work.
Action: Originate
Channel: SIP/$channeltocall/$extension
Exten: s
Context: test
Priority: 1
Async: yes
Codecs: alaw

Thanks
Jonh

8

You are going to need to provide debugging traces to work out what is conflicting with direct media. You’ll need the SDP exchange, for a start.

(Thinking about it, the Originate codecs are more relevant to pass through. I think the codecs will sort themselves out when the call connects, so shouldn’t inhibit direct media, so I think you have another contra-indication to direct media.)

9

[quote=“david55”]You are going to need to provide debugging traces to work out what is conflicting with direct media. You’ll need the SDP exchange, for a start.

(Thinking about it, the Originate codecs are more relevant to pass through. I think the codecs will sort themselves out when the call connects, so shouldn’t inhibit direct media, so I think you have another contra-indication to direct media.)[/quote]

I have a news.
Since I rebooted the server, the RTP traffic doesn’t go through asterisk and signaling still works fine.
Unfortuntely, when the call is placed, I don’t have ring back and once phones are bridged, they do not hear nothing.
I think also the VoIP provider have to permit directmedia.

How do I do change SDP paranmeters before the secod call is placed?

Jonh