The problem I'm having is that when I forward GV calls through the OBI212 to the Asterisk server voicemail SP3(100) the call is dropped after almost exactly 21 seconds of recorded voicemail. This happens during every call from GV that is forward through the OBI212 to Asterisk SIP server.
There is no issue when calls are forwarded to SP3(100) on the OBI212 LI port.
The asterisk debug console say "-- User hung up"
OBI212 syslog shows the following when calling in via Skype:
Oct 31 10:33:15 obi212 CCTL:NewCallOn Term 10[2] ->,100
Oct 31 10:33:15 obi212 RTP:DtmfTxMtd:1(1),0
Oct 31 10:33:15 obi212 RTP:DtmfTxMtd:1(1),0
Oct 31 10:33:15 obi212 RTP:Start->c0a80301:16402(80 0);0;0;0:0:0;0(46)
Oct 31 10:33:16 obi212 RTP:Start->4a7d271c:19305(80 0);0;0;0:0:0;0(44)
Oct 31 10:33:16 obi212 RTP:Set actpass 2
Oct 31 10:33:16 obi212 DTLS:Setup active
Oct 31 10:33:16 obi212 DTLS:Handshake Success
Oct 31 10:33:17 obi212 RTP:PeerRflxAddr=4a7d271c:19305
Oct 31 10:33:20 obi212 OB==>CRLFCRLF
Oct 31 10:33:20 obi212 OB:<==CRLF
Oct 31 10:33:20 obi212 OB:<==CRLF 1
Oct 31 10:33:21 obi212 RTCP:RxPkt[100]=200<--c0a80301:16403
Oct 31 10:33:26 obi212 RTCP:RxPkt[100]=200<--c0a80301:16403
Oct 31 10:33:34 obi212 RTCP:RxPkt[100]=200<--c0a80301:16403
Oct 31 10:33:36 obi212 Prd:SrvName=_sip._udp.callcentric.com; n=7
Oct 31 10:33:36 obi212 PRD:NOPriFbToTry
Oct 31 10:33:36 obi212 RTCP:RxPkt[80]=201<--c0a80301:16403
Oct 31 10:33:41 obi212 RTCP:RxPkt[80]=201<--c0a80301:16403
Oct 31 10:33:46 obi212 RTCP:RxPkt[80]=201<--c0a80301:16403
Oct 31 10:33:48 obi212 RTP:Del Channel
Oct 31 10:33:48 obi212 RTP:Del Channel
syslog shows the above plus the following from OBI202 when calling in via the OBI202:
Oct 31 11:02:01 obi202 RTP:Del Channel
Oct 31 11:02:04 obi202 [SLIC]: HOOK-FLASH early detected: 56055203
Oct 31 11:02:05 obi202 [SLIC]: TRUE HOOK-FLASH detected: 510, 0
Oct 31 11:02:05 obi202 [SLIC]:Slic#0 HOOK FLASH
OBI212 config.
SP1 GV primary outgoing line
SP2 GV
SP2 Asterisk (local) SIP server
SP4 Callcenteric
LI1 Verizon
All configs are standard for outgoing calls. SP1, SP2, LI1 incoming calls are forwarded to SP3(100).
SP1 ->Voice Services -> CallForwardUnconditionalNumber sp3(100)
SP1 ->ITSP Profile A -> SIP -> X_SpoofCallerID CHECKED
SP2 ->Voice Services -> X_InboundCallRoute sp3(100)
SP2 ->ITSP Profile A -> SIP -> X_SpoofCallerID CHECKED
SP3 -> Asterisk SIP Connection (local network)(Standard config as per OBItalk gw)
SP4 -> Callcenteric all standard configs for incoming and outgoing calls as per OBItalk gw
LI1 -> Physical Interfaces -> LINE Port -> CallForwardOnNoAnswerNumber sp3(100) after 5 rings
Since asterisk voicemail worked perfectly when calling extension directly from the OBI202. OBI212 and Zoiper 5,
but not when calls from google voice where being routed via OBI device to asterisk voicemail.
I wasn't sure where the problem was. It didn't help matters that I was also fighting an HTTP DDOS attack (What Fun, NOT).
The solution is a flag in asterisk.conf (transmit_silence = yes).
It took some google searching but I found the solution and explanation at the following URL.
https://thecomputerperson.wordpress.com/2017/05/10/asterisk-voicemail-hangs-up-on-callers-after-a-few-seconds/ (https://thecomputerperson.wordpress.com/2017/05/10/asterisk-voicemail-hangs-up-on-callers-after-a-few-seconds/)
I'm not sure how this setting will effect my landline to voicemail without a CPC (open loop disconnect) signal from verizon, Which I had working perfectly.
Just tested my landline to asterisk voicemail and it seemed to work fine. Asterisk hung up the landline after 10 seconds of silence and even cut the 10 seconds of silence out of the voicemail.