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Author Topic: Problem forwarding GV call through OBI212 [SOLVED]  (Read 1982 times)
dboling
Jr. Member
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Posts: 74



« on: October 31, 2018, 09:05:26 am »

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


« Last Edit: November 01, 2018, 02:14:08 am by dboling » Logged

-Diane
dboling
Jr. Member
**
Posts: 74



« Reply #1 on: November 01, 2018, 02:12:44 am »

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/

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.
« Last Edit: November 01, 2018, 02:58:31 am by dboling » Logged

-Diane
dboling
Jr. Member
**
Posts: 74



« Reply #2 on: November 01, 2018, 02:44:01 am »

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. 
Logged

-Diane
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