I'm running PIAF 13-13.10 on Centos 7. I setup a Signalwire Trunk and outbound calling works beautifully. Inbound goes to the black hole and it won't go the the destination extension.
I'm getting that pesky message:
WARNING[18212][C-00000003]: chan_sip.c:10395 process_sdp: Declining non-primary audio stream: audio 14028 RTP/AVP 0 9 101 13
SDP:
v=0
o=SignalWire-STACK 1562252741 1562252742 IN IP4 178.128.235.231
s=SignalWire-STACK
c=IN IP4 178.128.235.231
t=0 0
m=audio 14028 RTP/SAVP 0 9 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:slMtw6v7B1k6o52HkuSJBtcEVJGOeFpOicMRM6Ul
a=crypto:6 AES_256_CM_HMAC_SHA1_32 inline:4hSKRAZmN0QIRV/rH0u6QHJMLriA7ICJO2O70paBBxAee8sja7q3BnN94KFjzw==
a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:1bYBfXmOWaLgR1yI/HXTKxV40N8VaJAGjBzJuIVv
a=ptime:20
m=audio 14028 RTP/AVP 0 9 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20
I followed this thread for my configurations.
Incoming Trunk:
username:[email protected]/callerid
sip_custom_post.conf:
[signalwire1](signalwire);
host=sip.signalwire.com
[signalwire2](signalwire);
host=104.248.150.114
[signalwire3](signalwire);
host=178.128.235.81
[signalwire4](signalwire);
host=188.166.126.7
[signalwire5](signalwire);
host=159.65.244.171
[signalwire6](signalwire);
host=104.248.176.184
[signalwire7](signalwire);
host=167.99.198.84
Asterisk SIP Settings:
Allow Anonymous SIP Calls = Yes
Allow SIP Guests = Yes
***(I followed this post config. Do these two need to be set to Yes?)***
Enable TLS = Yes
SSL Method = tlsv1 (Still failed set to sslv2 and sslv3)
I'm getting that pesky message:
WARNING[18212][C-00000003]: chan_sip.c:10395 process_sdp: Declining non-primary audio stream: audio 14028 RTP/AVP 0 9 101 13
SDP:
v=0
o=SignalWire-STACK 1562252741 1562252742 IN IP4 178.128.235.231
s=SignalWire-STACK
c=IN IP4 178.128.235.231
t=0 0
m=audio 14028 RTP/SAVP 0 9 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:slMtw6v7B1k6o52HkuSJBtcEVJGOeFpOicMRM6Ul
a=crypto:6 AES_256_CM_HMAC_SHA1_32 inline:4hSKRAZmN0QIRV/rH0u6QHJMLriA7ICJO2O70paBBxAee8sja7q3BnN94KFjzw==
a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:1bYBfXmOWaLgR1yI/HXTKxV40N8VaJAGjBzJuIVv
a=ptime:20
m=audio 14028 RTP/AVP 0 9 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20
I followed this thread for my configurations.
Incoming Trunk:
username:[email protected]/callerid
sip_custom_post.conf:
[signalwire1](signalwire);
host=sip.signalwire.com
[signalwire2](signalwire);
host=104.248.150.114
[signalwire3](signalwire);
host=178.128.235.81
[signalwire4](signalwire);
host=188.166.126.7
[signalwire5](signalwire);
host=159.65.244.171
[signalwire6](signalwire);
host=104.248.176.184
[signalwire7](signalwire);
host=167.99.198.84
Asterisk SIP Settings:
Allow Anonymous SIP Calls = Yes
Allow SIP Guests = Yes
***(I followed this post config. Do these two need to be set to Yes?)***
Enable TLS = Yes
SSL Method = tlsv1 (Still failed set to sslv2 and sslv3)