Signalwire Trunk Inbound Not Working. Suggestions?

m2sdad71

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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)
 

m2sdad71

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Ok, progress.

Inbound calling connects when I initiate the call from my mobile to a softphone client. But the audio sounds underwater on the initiating end. The receiving end's audio sounds fine. When I disconnect the softphone so the call goes straight to voicemail, the voicemail greeting message is clear. I'm still getting this message and I'm speculating that it's related to the underwater audio:

WARNING[18212][C-0000000b]: chan_sip.c:10395 process_sdp: Declining non-primary audio stream: audio 18924 RTP/AVP 0 9 101 13

I set Anonymous SIP calls and Allow SIP Guests both to No, so I'm happy about that.

Any suggestions?
 

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