Comments (3)
Not sure I follow what you want to do. When you make an outbound call, you are sending an INVITE....the only way to truly redirect a call (if by redirect sending something like a 302 Moved) is when you are receiving an INVITE, not sending it...
You can certainly make an outbound call and, on answer, put the called party on hold and then connect them to another user / endpoint (by sending a second INVITE). Are you talking about something like that?
from drachtio-srf.
from drachtio-srf.
I'm still not sure I understand what you want.
From the asterisk documentation "The Asterisk Manager Interface (AMI) allows a client program to connect to an Asterisk instance and issue commands or read events over a TCP/IP stream"
This is exactly how drachtio-srf works: you connect to a drachtio server, receive an incoming call event and decide what to do with it.
Have you looked at any of the examples here? Are any of them close to what you want to do?
Perhaps what you want is to use outbound connections, which lets you receive an inbound call via a web app and then decide which drachtio-srf app should handle it.
I could certainly provide an example if I understood what you wanted to do..
from drachtio-srf.
Related Issues (20)
- PRACK interoperability not working HOT 3
- B2BUA 200 response missing SDP HOT 4
- OPTIONS request that causes a problem HOT 4
- Upgrade UUID to version 8 to avoid warnings for 3.4.0 HOT 2
- Broken links in https://drachtio.org/api HOT 1
- 407 response with re-Invite HOT 1
- ON Bye message in UAC/UAS , URl is replaced as placeholder and i am not getting bye message on the client HOT 1
- How can Srf client know about the loss of TCP/WS connection from SIP client to Drachtio server HOT 26
- how to place a call with asterisk HOT 3
- How to set SIP transport for createUac HOT 8
- SipError: Sip non-success response: 408 HOT 4
- createB2BUA can forward UAS Cancel request to UAC Cancel request.
- Can I get the headers of received CANCEL? HOT 17
- Do both Drachtio and SRF support SIP REPLACE?
- Public IP in Contact Header in #createUAS HOT 2
- Request-Line change request in SRF HOT 3
- Server error: unable to find dialog for dialog id provided HOT 1
- Getting periodic 486 and 503 errors HOT 3
- Srf types not updated correctly sometime after 4.5.23 HOT 3
- Capture all SIP messages in SIP Dialogs HOT 2
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