What is happening is when a call comes in on the main line and the caller dials the extension the caller can hear nothing but the call receiver can hear the caller and speak. Also If I dial direct to the extension audio works perfect. It only happens when the call is transferred
Insufficient information. For a start there are many different ways in which transfers can be done.
Here is the Debug
[code]—
Scheduling destruction of SIP dialog ‘K5zIElTdR7dddcjn2TcVvuEc9YfbXMlI’ in 32000 ms (Method: REGISTER)
<— SIP read from UDP:64.170.xxx.xxx:29377 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.170.xxx.xxx:28377;rport=28377;received=64.170.xxx.xxx;branch=z9hG4bK6fe1bc74
Call-ID: 7159e80355a9149e298319dd2f9721ba@64.170.xxx.xxx
From: “asterisk” sip:asterisk@64.170.xxx.xxx;tag=as6b90e63e
To: sip:417@64.170.xxx.xxx;ob;tag=z9hG4bK6fe1bc74
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: Digium D50 1_4_2_0_63880
Content-Type: application/sdp
Content-Length: 425
v=0
o=- 113959953 113959953 IN IP4 64.170.xxx.xxx
s=digphn
c=IN IP4 64.170.98.168
t=0 0
m=audio 4000 RTP/AVP 0 8 9 111 18 58 118 58 96
a=rtcp:4001 IN IP4 64.170.xxx.xxx
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:18 G729/8000
a=rtpmap:58 L16/16000
a=rtpmap:118 L16/8000
a=rtpmap:58 L16-256/16000
a=sendrecv
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
<------------->
— (13 headers 18 lines) —
Really destroying SIP dialog ‘7159e80355a9149e298319dd2f9721ba@64.170.98.15:28377’ Method: OPTIONS
<— SIP read from UDP:64.170.xxx.xxx —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.170.xxx.xxx;rport=xxxx;received=64.170.xxx.xxx;branch=z9hG4bK74dbeaa3
Call-ID: 5251d21704fdeece0822bcd5141db0e7@64.170.xxx.xxx:
From: “asterisk” sip:asterisk@64.170.xxx.xxx;tag=as2d8fb84d
To: sip:417@64.170.xxx.xxx;ob;tag=z9hG4bK74dbeaa3
CSeq: 102 NOTIFY
Content-Length: 0
<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘5251d21704fdeece0822bcd5141db0e7@64.170.xxx.xxx’ Method: NOTIFY
Reliably Transmitting (NAT) to 162.243.35.55:44121:
OPTIONS sip:418@10.0.1.3:44121;rinstance=11C2CFFD SIP/2.0
Via: SIP/2.0/UDP 64.170.xxx.xxx;branch=z9hG4bK00d086c4;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@64.170.xxx.xxx;tag=as62d6e907
To: sip:418@10.0.1.3:44121;rinstance=11C2CFFD
Contact: sip:asterisk@64.170.xxx.xxx
Call-ID: 36b68090474f523768743ac86fac96cf@64.170.98.15:28377
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.13.0
Date: Sun, 14 Dec 2014 20:13:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0
<— SIP read from UDP:162.243.35.55:44121 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.170.98.15:28377;branch=z9hG4bK00d086c4;rport=28377;received=64.170.98.15
Contact: sip:418@10.0.1.3:44121;rinstance=11C2CFFD
From: “asterisk” sip:asterisk@64.170.xxx.xxx;tag=as62d6e907
Call-ID: 36b68090474f523768743ac86fac96cf@64.170.xxx.xxx
CSeq: 102 OPTIONS
To: sip:418@10.0.1.3:44121;rinstance=11C2CFFD
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
Accept: application/sdp
Content-Length: 0
<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘36b68090474f523768743ac86fac96cf@64.170.98.15:28377’ Method: OPTIONS
Reliably Transmitting (NAT) to 71.198.111.249:29377:
OPTIONS sip:419@10.0.0.24:29377 SIP/2.0
Via: SIP/2.0/UDP 64.170.xxx.xxx;branch=z9hG4bK765708b6;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@64.170.xxx.xxx;tag=as1b0b4477
To: sip:419@10.0.0.24:29377
Contact: sip:asterisk@64.170.xxx.xxx
Call-ID: 747a88d6679f14784fbce1f242fcfd05@64.170.98.15:28377
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.13.0
Date: Sun, 14 Dec 2014 20:13:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0
<— SIP read from UDP:71.198.111.249:29377 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.170.xxx.xxx;branch=z9hG4bK765708b6;rport
From: “asterisk” sip:asterisk@64.170.xxx.xxx;tag=as1b0b4477
To: sip:419@10.0.0.24:29377;tag=3209036821
Call-ID: 747a88d6679f14784fbce1f242fcfd05@64.170.xxx.xxx
CSeq: 102 OPTIONS
User-Agent: Yealink SIP-W52P 25.73.0.20
Content-Length: 0[/code]