Hello Obi experts,
here is a follow-up question after I used an Obi202 to obify my IP and IP dect phones using the link:
https://voxilla.com/2011/02/15/how-to-obify-your-ip-phone-no-asterisk-involved/
With my IP Phone and IP Dect connected to GV using Obi202 as a proxy per the link above, while calling/receiving GV calls mostly work, I seem to be facing 603 media inactivity issues for certain situations e.g. if the called party's VM does not answer after 4 rings or putting the call on hold for more than 25 sec or if called party doesn't have a VM setup yet etc. The same issues don't happen if using an analog phone connected to one of Obi's phone ports.
Given the above behavior, is there a setting I missed when Obifying my IP phones which leads to the above mentioned errors?
Thanks!
Review the 2nd post here
https://www.dslreports.com/forum/r32067996-GVSIP-Asterisk-Freepbx-503-error-on-voice-mail-handoff
Yes, I saw that post previously but not sure how that is related to my issue. I mean, that is applicable to asterisk but as I mentioned, I use obi202 as a proxy with with my IP phones without anything to do with any kind of pbx.
Your symptoms relate closely to the first issue described in that thread pertaining to unanswered calls being handed off to voicemail getting dropped.
I haven't looked too closely at the obi sip/voice services setting options, but perhaps the solution in that thread might guide you to a similar setting in the obi.
@azrobert, are you experiencing similar issues with unanswered calls to gv numbers when using the obi as proxy?
I guess my point was I didn't see anything remotely similar in obi dial plan which would be considered equivalent to the 'r' option on asterisk.
Quote from: HMishra on August 05, 2018, 09:34:54 AM
I guess my point was I didn't see anything remotely similar in obi dial plan which would be considered equivalent to the 'r' option on asterisk.
I'll look closer at the different options when I have more time. I'm actually doing the reverse here, using asterisk/freepbx as a proxy for the obi for certain types of calls.
Also, I that link in the OP is from 2011. It's possible some other changes need to be done as well to make it work today?
I recall seeing very recent posts on here by azrobert describing how to the same thing.
*fingers crossed* Hope there is a fix.
I'm throwing this out completely to far left field, but worth a shot.
Service Providers -> ITSP Profile {which ever one you're using for the gv proxy} -> SIP -> X_SpoofCallerID: Checked
Apply then reboot, retest.
Yep, it already is, per instructions in the OP. Thanks for continued help!
Try the suggestion by stewart and ronr in this thread:
http://www.obitalk.com/forum/index.php?topic=2235.0
Note the modifications ronr makes in the 7th or 8th post.
Also, this goes in the Phoneport's, OutboundCallRoute field.
Edit what's there and adjust the {(<**1:>(Msp1)):sp1} as shown. Change the 1 to the respective sp slot, in all 3 places.
For the phone #, try a 10 digit or 11 digital number that's *NOT* your gv #.
When doing your testing, Dial **1 (or **2, etc) rather than the phone number directly. This ensures that CID number is sent. If this fixes the problem then we can figure out how to pass it as default.
Again, i'm only going down this route because that's what solved the 2nd issue in that dslr post - where dialing one's own # resulted in a 603 decline error.
I am kinda lost here. Here is what exists on my current OutboundCallRoute for both phoneports:
{911:sp3},{933:sp3},{([1-9]x?*(Mpli)):pp},{(<##:>):li},{(<#:>):ph2},{(<**70:>(Mli)):li},{(<**82:>(Mbt2)):bt2},{(<**81:>(Mbt)):bt},{(<**8:>(Mbt)):bt},{**0:aa},{***:aa2},{(<**1:>(Msp1)):sp1},{(<**2:>(Msp2)):sp2},{(<**3:>(Msp3)):sp3},{(<**4:>(Msp4)):sp4},{(<**9:>(Mpp)):pp},{(Mpli):pli}
I use SP2 as GV proxy, SP1 being the actual configured GV connection and SP3 is for my CallCentric connection which I use for 911. That said, what should I change? I guess I am confused about how caller-id comes into picture.
Thanks!
Testing a theory based on the that dslr thread that it's not forwarding to vm correctly because a number equaling your gv # is getting passed as the caller id which somehow causes issues.
Theory is to pass on a different caller id when placing the call, which, according to the dslr thread, in asterisk fixed the issue.
So, you would update
{911:sp3},{933:sp3},{([1-9]x?*(Mpli)):pp},{(<##:>):li},{(<#:>):ph2},{(<**70:>(Mli)):li},{(<**82:>(Mbt2)):bt2},{(<**81:>(Mbt)):bt},{(<**8:>(Mbt)):bt},{**0:aa},{***:aa2},{(<**1:>(Msp1)):sp1},{(<**2:>(Msp2)):sp2},{(<**3:>(Msp3)):sp3},{(<**4:>(Msp4)):sp4},{(<**9:>(Mpp)):pp},{(Mpli):pli}
with
{911:sp3},{933:sp3},{([1-9]x?*(Mpli)):pp},{(<##:>):li},{(<#:>):ph2},{(<**70:>(Mli)):li},{(<**82:>(Mbt2)):bt2},{(<**81:>(Mbt)):bt},{(<**8:>(Mbt)):bt},{**0:aa},{***:aa2},{(<**1:>(Msp1)):sp1(12341234567>$2)},{(<**2:>(Msp2)):sp2},{(<**3:>(Msp3)):sp3},{(<**4:>(Msp4)):sp4},{(<**9:>(Mpp)):pp},{(Mpli):pli}
When placing your test call, dial **1{number}, where number is one of those numbers that created issues for you in the past when unanswered.
Hmmm...After above change, I get a busy tone when dialing the same # that I entered on the OutboundCallRoute.
I don't follow. You're dialing a real phone number or 12341234567 ? If the latter, why? What happens with a real number that caused issues in the past?
I substituted 12341234567 with my real phone# (my cell#) both in the outboundcallroute and while dialing e.g. **1 + <number> and got a busy tone. It wasn't recorded on my Obi202 call history either so not sure what happened.
You missed my point.. The number that should go in there was the 12341234567. It's not a place holder, but literally that number.
Seems that I did! My bad....
Will make changes when I get back home in the evening and report back here.
I tried with your suggested outboundcallroute and dialing **1 + <my cell phone#> with the same result......busy tone.
This is turning out to be an exercise in futility. I think I will probably spend some time setting up a Asterisk/freepbx instance since I have a spare raspberry pi3 anyway. *fingers crossed*
Let me ask, when you're dialing another gv number are you hearing ring back tones or silence? If silence then stop reading here.
If ringback tones, see if there's a setting in your voip phone that disables that. That was part of the issue with the vm handoff in that dslr thread. Getting rid of the asterisk generated ring tones resulted in silence during the 'ringing' stage, the actual hand off to vm was successful.
Passing one's own gv # as cid when calling one's own gv number resulted in dropped calls. Passing any number other than one's own worked to reach the gv vm system to listen to your own messages and/or place a call by pressing 2.
I do hear ring back tone when I call my own GV# or my cell#.
I have tried everything that I can think of or gleaned from various forums/threads to set a different caller-id other than my own GV# including your previous suggestion and nothing seemed have worked. Not sure what else I can try.
Question then is, are those ring tones generated by the number you're calling, or by the sip phone itself. I suspect the latter, and this is what's causing the problem.
What's the make/model of sip phones in question?
Both my phones are from Grandstream, one is GXP2170 (Desk IP Phone) and the other is DP750/DP720 (One IP Dect base unit and 6 handsets).
I couldn't find any relevant info for either.
Look through the phone settings for anything to do with dial options or ringback tones.
Also, perhaps set up a soft phone on the pc to connect to the obi to see how it handles such calls.
The problem sounds similar to the one I am facing:
Outgoing calls with Zoiper (503 Service Unavailable / 603 media inactivity) (http://www.obitalk.com/forum/index.php?topic=14452.msg92188#msg92188)
I am told this is due to a bug in the firmware.