sangsoo
Member
Hello.
I'm testing SIP functionality, and when I make a call, it returns "Busy."
- https://docs.flashphoner.com/display/WCS52EN/A+call+between+two+browsers+made+via+the+SIP+server
- The "Without an external SIP server. SIP and RTP media are processed by WCS" section in the documentation works fine.
I configured andrius/asterisk as a Docker image on a different server (not WCS) and ran it in network-hosted mode.
- https://hub.docker.com/r/andrius/asterisk
My wcs flashphoner.properties:
dtmf=RFC2833
-----Version info-----
wcs_version=5.3.126-4162ea876dc627c303f32b785af69ac5304261c5
wcs_client_version=2.0.259-485b3fbffa4fd4abe88fbdc455691a05bbf79025
I'm trying to connect a call using two extension(internal) numbers (7003 and 7006).
The registration succeeds, but when I answer the call, it immediately disconnects with a 486 error message.
I've attached the asterisk log (486.txt). Can you help me figure out what the problem is?
Also, what configurations are required for the next two steps in the Operation flowchart? I'd like to check each flow.
1. SIP server as a proxy server to transfer calls and RTP media
2. SIP server as a server to transfer calls only
Please answer. Thank you.
I'm testing SIP functionality, and when I make a call, it returns "Busy."
- https://docs.flashphoner.com/display/WCS52EN/A+call+between+two+browsers+made+via+the+SIP+server
- The "Without an external SIP server. SIP and RTP media are processed by WCS" section in the documentation works fine.
I configured andrius/asterisk as a Docker image on a different server (not WCS) and ran it in network-hosted mode.
- https://hub.docker.com/r/andrius/asterisk
My wcs flashphoner.properties:
dtmf=RFC2833
-----Version info-----
wcs_version=5.3.126-4162ea876dc627c303f32b785af69ac5304261c5
wcs_client_version=2.0.259-485b3fbffa4fd4abe88fbdc455691a05bbf79025
I'm trying to connect a call using two extension(internal) numbers (7003 and 7006).
The registration succeeds, but when I answer the call, it immediately disconnects with a 486 error message.
I've attached the asterisk log (486.txt). Can you help me figure out what the problem is?
Also, what configurations are required for the next two steps in the Operation flowchart? I'd like to check each flow.
1. SIP server as a proxy server to transfer calls and RTP media
2. SIP server as a server to transfer calls only
Please answer. Thank you.
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