I have a SIP trunk set up with Twilio for outbound calls. Twilio-FreePBX and then my test device is the simple X-Lite from CounterPath.
I can make an outgoing call from X-Lite. My cell phone rings and I can pick up. But that's it, there is no audio transmitted and the call hangs up on it's own after a handful of seconds.
This is the error I receive from the Asterisk log inside my FreePBX server:
[2017-02-17 14:41:46] WARNING[1996] chan_sip.c: Retransmission timeout reached on transmission 83369MWU3MmY5MWZiZWZkODJmYjc3ZWEzMWI5ZmQzMTQ1NWQ for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[2017-02-17 14:41:46] WARNING[1996] chan_sip.c: Hanging up call 83369MWU3MmY5MWZiZWZkODJmYjc3ZWEzMWI5ZmQzMTQ1NWQ - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
The call is routing through my Twilio account, I can see it in the logs there. It registers as complete.
I've turned the FreePBX firewall on and added trusted IP's
The full Asterisk debug log:
<------------->
[2017-02-17 15:18:58] VERBOSE[1996] chan_sip.c: --- (12 headers 12 lines) ---
[2017-02-17 15:18:58] VERBOSE[1996][C-00000004] chan_sip.c: Found RTP audio format 0
[2017-02-17 15:18:58] VERBOSE[1996][C-00000004] chan_sip.c: Found RTP audio format 101
[2017-02-17 15:18:58] VERBOSE[1996][C-00000004] chan_sip.c: Found audio description format PCMU for ID 0
[2017-02-17 15:18:58] VERBOSE[1996][C-00000004] chan_sip.c: Found audio description format telephone-event for ID 101
[2017-02-17 15:18:58] VERBOSE[1996][C-00000004] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|g726), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
[2017-02-17 15:18:58] VERBOSE[1996][C-00000004] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2017-02-17 15:18:58] VERBOSE[1996][C-00000004] chan_sip.c: Peer audio RTP is at port 54.172.61.111:12510
[2017-02-17 15:18:58] VERBOSE[1996][C-00000004] sip/route.c: sip_route_dump: route/path hop: <sip:54.172.60.2:5060;lr;ftag=as7484893d;twnat=sip:52.90.124.243:5060>
[2017-02-17 15:18:58] VERBOSE[1996][C-00000004] chan_sip.c: Transmitting (NAT) to 54.172.60.2:5060:
ACK sip:172.18.7.119:5060 SIP/2.0
Via: SIP/2.0/UDP 172.**.**.***:5060;branch=z9hG4bK2b6c05a0;rport
Route: <sip:54.172.60.2:5060;lr;ftag=as7484893d;twnat=sip:52.90.124.243:5060>
Max-Forwards: 70
From: <sip:PHIL@172.**.**.***>;tag=as7484893d
To: <sip:+18566492240@********.pstn.twilio.com>;tag=77864250_6772d868_655d5c53-0b14-4aa5-8bd5-d8f83501d26c
Contact: <sip:PHIL@172.**.**.***:5060>
Call-ID: 664a272c08f7af0543b2bac950391d32@172.**.**.***:5060
CSeq: 103 ACK
User-Agent: FPBX-13.0.190.7(13.12.2)
Content-Length: 0
---
[2017-02-17 15:18:58] VERBOSE[8613][C-00000004] app_dial.c: SIP/Twilio Trunk-00000009 answered SIP/808-00000008
[2017-02-17 15:18:58] VERBOSE[8613][C-00000004] chan_sip.c: Audio is at 12824
[2017-02-17 15:18:58] VERBOSE[8613][C-00000004] chan_sip.c: Adding codec ulaw to SDP
[2017-02-17 15:18:58] VERBOSE[8613][C-00000004] chan_sip.c: Adding codec alaw to SDP
[2017-02-17 15:18:58] VERBOSE[8613][C-00000004] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2017-02-17 15:18:58] VERBOSE[8613][C-00000004] chan_sip.c:
<--- Reliably Transmitting (NAT) to 73.81.116.96:35304 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 73.81.116.96:35304;branch=z9hG4bK-524287-1---d5cf0232413c0269;received=73.81.116.96;rport=35304
From: "Phil"<sip:808@52.90.124.243>;tag=82678409
To: <sip:+18566492240@52.90.124.243>;tag=as08c320c9
Call-ID: 83369ZWQ2OTI2MjRkNGE3MTdlYmM5MjYxM2Q0ZDIwOWVhYTM
CSeq: 2 INVITE
Server: FPBX-13.0.190.7(13.12.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:+18566492240@172.**.**.***:5060>
Content-Type: application/sdp
Content-Length: 278
v=0
o=root 1504626483 1504626483 IN IP4 172.**.**.***
s=Asterisk PBX 13.12.2
c=IN IP4 172.**.**.***
t=0 0
m=audio 12824 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<------------>
[2017-02-17 15:18:58] VERBOSE[8637][C-00000004] bridge_channel.c: Channel SIP/Twilio Trunk-00000009 joined 'simple_bridge' basic-bridge <e035a287-8fa3-4291-a3ad-927bca1407e0>
[2017-02-17 15:18:58] VERBOSE[8613][C-00000004] bridge_channel.c: Channel SIP/808-00000008 joined 'simple_bridge' basic-bridge <e035a287-8fa3-4291-a3ad-927bca1407e0>
[2017-02-17 15:18:58] VERBOSE[1996] chan_sip.c: Retransmitting #1 (NAT) to 73.81.116.96:35304:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 73.81.116.96:35304;branch=z9hG4bK-524287-1---d5cf0232413c0269;received=73.81.116.96;rport=35304
From: "Phil"<sip:808@52.90.124.243>;tag=82678409
To: <sip:+18566492240@52.90.124.243>;tag=as08c320c9
Call-ID: 83369ZWQ2OTI2MjRkNGE3MTdlYmM5MjYxM2Q0ZDIwOWVhYTM
CSeq: 2 INVITE
Server: FPBX-13.0.190.7(13.12.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:+18566492240@172.**.**.***:5060>
Content-Type: application/sdp
Content-Length: 278
v=0
o=root 1504626483 1504626483 IN IP4 172.**.**.***
s=Asterisk PBX 13.12.2
c=IN IP4 172.**.**.***
t=0 0
m=audio 12824 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
The Retransmitting goes on for 6 tries before killing the connection.
Any help would be great. Thanks in advance!
