Asterisk 12/PJSIP - Temporary failure in sending Request

Asterisk Asterisk Support
1

I am having an issue with setting up a call between two devices using Asterisk 12 and PJSIP.

My problem is extension 100 can dial 101 no problem. However, when 101 dials 100 I get this error:


[Nov  1 08:43:28] WARNING[3368]: pjsip:0 <?>: 	tsx0x7ffe08006 Temporary failure in sending Request msg INVITE/cseq=12255 (tdta0x2b4bd10), will try next server. Err=120111 (Connection refused)
    -- Channel PJSIP/sip101_abcdefg-0000000c left 'native_rtp' basic-bridge <b4819b1e-92e0-4d59-b09b-9bfc6546c04a>
    -- Channel PJSIP/sip100_abcdefg-0000000d left 'native_rtp' basic-bridge <b4819b1e-92e0-4d59-b09b-9bfc6546c04a>

I came across this article but not sure how it needs to be approached in relation to the Asterisk config: trac.pjsip.org/repos/wiki/Using_SIP_TCP

Here is the full log output for both events.

100 -> 101 (Working)

    -- Executing [101@abcdefg:1] Dial("PJSIP/sip100_abcdefg-00000032", "PJSIP/sip101_abcdefg,25")
    -- Called PJSIP/sip101_abcdefg
    -- PJSIP/sip101_abcdefg-00000033 is ringing
    -- PJSIP/sip101_abcdefg-00000033 answered PJSIP/sip100_abcdefg-00000032
    -- Channel PJSIP/sip100_abcdefg-00000032 joined 'simple_bridge' basic-bridge <cbe9c1c8-7b7d-40f7-a917-5cf17f6ef655>
    -- Channel PJSIP/sip101_abcdefg-00000033 joined 'simple_bridge' basic-bridge <cbe9c1c8-7b7d-40f7-a917-5cf17f6ef655>
       > Bridge cbe9c1c8-7b7d-40f7-a917-5cf17f6ef655: switching from simple_bridge technology to native_rtp
[Nov  1 09:06:36] ERROR[3368]: pjsip:0 <?>: 	tcpc0x7ffe0808 TCP connect() error: Connection refused [code=120111]
[Nov  1 09:06:36] WARNING[3368]: pjsip:0 <?>: 	  tsx0x287ac18 Temporary failure in sending Request msg INVITE/cseq=6722 (tdta0x289d080), will try next server. Err=120111 (Connection refused)
       > 0x7ffe08074050 -- Probation passed - setting RTP source address to 192.168.3.169:5074
       > 0x7ffe08064b40 -- Probation passed - setting RTP source address to 192.168.3.121:5004

101 -> 100 (Not-Working)

    -- Executing [100@abcdefg:1] Dial("PJSIP/sip101_abcdefg-00000040", "PJSIP/sip100_abcdefg,25")
    -- Called PJSIP/sip100_abcdefg
    -- PJSIP/sip100_abcdefg-00000041 is ringing
[Nov  1 09:09:27] WARNING[3560]: res_pjsip_sdp_rtp.c:234 set_caps: No joint capabilities between our configuration((gsm|ulaw|alaw|g729)) and incoming SDP((nothing))
    -- PJSIP/sip100_abcdefg-00000041 answered PJSIP/sip101_abcdefg-00000040
    -- Channel PJSIP/sip101_abcdefg-00000040 joined 'simple_bridge' basic-bridge <a127c3da-71c6-423f-8659-54b69be27fd6>
    -- Channel PJSIP/sip100_abcdefg-00000041 joined 'simple_bridge' basic-bridge <a127c3da-71c6-423f-8659-54b69be27fd6>
       > Bridge a127c3da-71c6-423f-8659-54b69be27fd6: switching from simple_bridge technology to native_rtp
       > 0x7ffe080556f0 -- Probation passed - setting RTP source address to 192.168.3.169:5102
[Nov  1 09:09:27] ERROR[3368]: pjsip:0 <?>: 	tcpc0x7ffe080a TCP connect() error: Connection refused [code=120111]
[Nov  1 09:09:27] WARNING[3368]: pjsip:0 <?>: 	  tsx0x2843198 Temporary failure in sending Request msg INVITE/cseq=21044 (tdta0x283cb90), will try next server. Err=120111 (Connection refused)
    -- Channel PJSIP/sip101_abcdefg-00000040 left 'native_rtp' basic-bridge <a127c3da-71c6-423f-8659-54b69be27fd6>
  == Spawn extension (abcdefg, 100, 1) exited non-zero on 'PJSIP/sip101_abcdefg-00000040'
    -- Channel PJSIP/sip100_abcdefg-00000041 left 'native_rtp' basic-bridge <a127c3da-71c6-423f-8659-54b69be27fd6>
2

I was able to get this going by setting the following in ps_endpoints:

allow = ulaw;alaw
direct_media = no
disable_direct_media_on_nat = yes