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Author Topic: SOLVED: Google Voice -> OBI 110 -> PIAF-Purple  (Read 19169 times)
DocM
Jr. Member
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Posts: 44


« on: April 16, 2013, 08:03:56 am »

I have a google voice connection that is connected via my OBI 110 to my PIAF Purple server. Recently, I was notified that my asterisk voicemails were cut off after ~ 22 seconds. This is the only problem I find with the Google Voice lines. I can reproduce the problem by calling the Google Voice line and being redirected to an extension's voicemail. I believe the real problems lies with the OBI110 and not the Google Voice lines. I believe that asterisk closes the connection if no sound is emitted from the PBX to the caller.

Via the FreePBX interface, I examined the Asterisk Logfiles for some test calls I've made to the PBX. The line that catches my attention is "app.c: No audio available on SIP/TRUNKxxx-xxx??"

I read that RTP settings may help solve this problem. I've tried altering the RTP settings in the OBI110 (both the PBX connection and Google Voice connection) to no avail. I don't know how to set the RTP settings for the OBI110 trunk in FreePBX or if its even possible.

I would be grateful for any help provided. Thanks in advance.

PBX in a Flash PURPLE Status Program
------------------------------------------------------------------------------
+-------------------SYSTEM INFORMATION *VERIFIED*---------------------+
Asterisk = ONLINE | Dahdi = ONLINE | MySQL = ONLINE
SSH = ONLINE | Apache = ONLINE | Iptables = ONLINE
Fail2ban = ONLINE | Internet = ONLINE | Ip6Tables = ONLINE
Disk Free = ADEQUATE| Mem Free = ADEQUATE| NTPD = ONLINE
SendMail = ONLINE | Samba = OFFLINE | Webmin = ONLINE
Ethernet0 = ONLINE | Ethernet1 = N/A | Wlan0 = N/A

PIAF Installed Version = 2.0.6.2 under *HARDWARE*
FreePBX Version = 2.9.0.12
Running Asterisk Version = 1.8.19.1
Asterisk Source Version = 1.8.19.1
Dahdi Source Version = 2.6.1+2.6.1
Libpri Source Version = 1.4.12
IP Address = xxx.xxx.xxx.xxx on eth0
Operating System = CentOS release 6.2 (Final)
Kernel Version = 2.6.32-220.7.1.el6.i686 - 32 Bit
Incredible PBX 3 Version = 3.0.7
+---------------------------------------------------------------------+
« Last Edit: April 29, 2013, 02:12:44 pm by DocM » Logged
DocM
Jr. Member
**
Posts: 44


« Reply #1 on: April 16, 2013, 05:40:10 pm »

I can't be the only one connecting GV to an asterisk server via an OBI110, right? I hope not... Any help would be appreciated. Even links to some clue of how I may solve this problem.
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QBZappy
Hero Member & Beta Tester
*****
Posts: 2322



« Reply #2 on: April 16, 2013, 06:39:24 pm »

This issue has come up before:
Have a look here:
Transferred Obi calls hang up exactly 20 seconds into leaving a GV voicemail
http://www.obitalk.com/forum/index.php?action=printpage;topic=3180.0

DocM
It seems you participated in the thread.
« Last Edit: April 16, 2013, 06:44:12 pm by QBZappy » Logged

Owner of the 1st OBi110/100 units in service in Canada & South America. 1st OBi202 on my street. 1st OBi1032 in Montreal.
DocM
Jr. Member
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Posts: 44


« Reply #3 on: April 18, 2013, 02:14:28 pm »

Thanks QBZappy for the reply. The problem is that the message isn't being recorded by Google Voice. The message is being recorded by my asterisk system (PIAF). The asterisk system works fine with OBI as long as the conversation is two way. However, when leaving a voicemail, the asterisk system should be patiently listening while recording the message but instead cuts off the line due to it considering the line is dead. I think I might be able to replicate this using the OBI110 with any service other than Google Voice but I haven't yet tried.
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QBZappy
Hero Member & Beta Tester
*****
Posts: 2322



« Reply #4 on: April 19, 2013, 06:13:37 pm »

DocM,

If you have come to the point that you are desperate and willing to try something/anything, then you might try something that azrobert found that affects audio. I don't see the connection but it doesn't hurt to try.

To fix audio problems do following on the OBi110.

Service Providers -> ITSP Profile B -> SIP -> X_DiscoverPublicAddress: unchecked
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Owner of the 1st OBi110/100 units in service in Canada & South America. 1st OBi202 on my street. 1st OBi1032 in Montreal.
DocM
Jr. Member
**
Posts: 44


« Reply #5 on: April 19, 2013, 07:49:20 pm »

That solution probably wouldn't apply to me since my two way audio is fine but PIAF doesn't detect a live connection. I will try it none-the-less, otherwise I'm going to have to dig through the PIAF code to implement a "fix".

Thanks again for your help QBZappy.
« Last Edit: April 19, 2013, 07:54:07 pm by DocM » Logged
RFord
Full Member
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Posts: 218


« Reply #6 on: April 20, 2013, 05:53:42 am »

DOCM:

Are you using GV setup natively on PIAF and the OBi110 as an extension registered with your PIAF Server?
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DocM
Jr. Member
**
Posts: 44


« Reply #7 on: April 22, 2013, 04:19:56 pm »

Thanks RFord for the reply. GV is setup natively on OBI110 and it is registered to PIAF. I did this due to problems GV had running on PIAF in the past. Not sure if this changed.
« Last Edit: April 23, 2013, 06:16:40 pm by DocM » Logged
DocM
Jr. Member
**
Posts: 44


« Reply #8 on: April 29, 2013, 02:12:15 pm »

Thank you QBZappy and RFord for your replies. After two weeks I found the solution.

To resolve Google voice voicemail interruption in PIAF:
1) go to asterisk.config via config edit in freepbx
2) uncomment [options] and transmit_silence_during_record = yes
3) restart asterisk

Explanation:
GV will cut call if it doesn't hear anything back within 20 seconds. I'm not sure why it didn't affect me in the past but it affects all current piafs (PURPLE and GREEN).
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