SOLVED In bound call results in partial invite without SDP

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Jul 15, 2021
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I had a setup where incoming calls used to hit the extensions without issue. I rebooted freeswitch few times as the toggling of ringback variable was not getting reflected unless freeswitch was restarted. Can some one let me know what did I mess up.

When an inbound call hits freeswitch, I transfer it to an extension. The call hits the dialplan and then freeswitch sends an invite like this without any codec information

2021-09-16 19:22:10.237854 [DEBUG] mod_sofia.c:93 sofia/internal/13@192.168.1.12:46962 SOFIA INIT 2021-09-16 19:22:10.237854 [DEBUG] sofia_glue.c:1618 sofia/internal/13@192.168.1.12:46962 sending invite version: 1.10.3 -release 32bit Local SDP: v=0 o=FreeSWITCH 1631767896 1631767897 IN IP4 192.168.1.2 s=FreeSWITCH c=IN IP4 192.168.1.2 t=0 0

The client responds with

recv 375 bytes from tcp/[192.168.1.12]:46962 at 19:22:10.536394: ------------------------------------------------------------------------ SIP/2.0 415 Unsupported Media Type Via: SIP/2.0/TCP 192.168.1.2;branch=z9hG4bK7Z73FZF7yKygj To: <sip:13@192.168.1.12:46962;rinstance=7139c2059776e62e;transport=tcp>;tag=d1684153
 
Jul 15, 2021
102
9
18
33
The following lines had disappeared from the XML in the inboundroutes, false alarm - all well now

<action application="set" data="domain_uuid=344f3a18-9840-4920-b0c3-4e205a05c18e" inline="true"/>
<action application="set" data="domain_name=domainname" inline="true"/>
 
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