SOLVED No incoming/outgoing calls with Obivoice

Todd Robinson

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Hey guys, I'd appreciate some help. I'm running PIAF 2.0.6.5 Green and am attempting to set up my Obivoice account with no success. I have an existing Google Voice account that works perfectly inbound and outbound.

Here's my trunk peer config:
Code:
host=sms.intelafone.com
fromdomain=sms.intelafone.com
username=xxxx
fromuser=xxxx
secret=xxxxxx
type=peer
disallow=all
allow=ulaw
insecure=port,invite
context=from-trunk
port=5060

The string I use for registration is: XXXX:[password]@sms.intelafone.com:5060/9199736245. My Obivoice customer portal shows the registration of my PBX and likewise Reports -> Asterisk Info -> Registries lists the Obivoice account.

I have an Inbound Route that includes all DIDs and all CIDs. When I make calls to the Obivoice number, there is no answer. It promptly hangs up. At this point nothing shows up in the Asterisk logs, although after tinkering with the router and temporarily forwarding ports 5060, 5060, and 10001-20000 did being showing results in the log that I did not capture but it appeared that anonymous calls were being rejected. I cannot reproduce that scenario now, however.

Outbound calling results in an all circuits are busy voice prompt.

Any suggestions? I've researched and tinkered about all I can and am at the end of my rope.
 

rossiv

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Did you allow ObiVoice's servers through the whitelist?
 

Todd Robinson

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Forgive my ignorance but which whitelist are you referring to? My hardware firewall? IPTables? Does FreePBX have one?

I have quite a bit of experience with managing devices and setting up call handling for our PBX at work but have never managed trunks before.
 

jeffmac

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If you are registering with Obivoice, then you shouldn't have to change the whitelist (which, by the way, is in iptables.)

But you really need to be able to see the logs to know that your inbound routes are matching the format Obivoice is sending. I don't use Obivoice at this point, so I can't make any suggestion on what the inbound route need to look like.

On possibility would be to allow anonymous SIP and try again to see if you get some log records. You should also verify the verbosity level that Asterisk is running with (I believe the default is 3). If you want it to be more verbose you can login to the asterisk command line interface with something like: asterisk -rvvvvv which should set the verbosity to 5, then you can log out of the CLI by entering quit. (The verbosity change will remain, in my experience, until Asterisk is recycled).

Jeff
 

wardmundy

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You don't need to allow anonymous SIP with Obivoice. Here's the Trunk setup that works for me. Then add an Inbound Route with your 10-digit phone number as your DID and point to an extension.
Code:
host=sms.intelafone.com
fromdomain=sms.intelafone.com
username=xxxx ; where xxxx is your acct #
fromuser=xxxx
secret=yourpassword
type=peer
port=5060
disallow=all
allow=ulaw
insecure=port,invite
context=from-trunk
 
Register string: xxxx:[email protected]/your10digitphone#
 

Todd Robinson

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A couple of updates, last night outbound started working without intervention. Happy happy joy joy. :)

So incoming remains a problem. Ward, I copied and pasted your config, changing the info pertaining to my account. It was close to what I had before, except port=5060 was the last item in the line. Now when I call in it will ring once and then hang up whereas before it wouldn't ring at all, just dead air and click. I already had an inbound route defining all DIDs and sending those calls to an extension. I did specify an additional route for just my Obivoice DID but it ends with the same result.

I turned on verbose logging as suggested. No luck getting anything there (checking FreePBX -> Reports -> Asterisk Logfiles -> full). There are no log events for my call to the Obivoice number. I don't understand where the break point is here...again, I'm new to SIP trunking. I would add that this PIAF install is new, only changes being the extensions, a ring group, GV trunk and an all DID/CID inbound route that does work for GV. Have I left any configuration options out? Asterisk SIP Settings? Totally grasping at straws here.
 

wardmundy

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What does the Asterisk CLI show during the incoming call? Should also be documented in /var/log/asterisk/full.
 

Todd Robinson

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Ward, the CLI doesn't show anything when I call the Obivoice number. The CLI displays all other actions in verbose detail but absolutely nothing when dialing the Obivoice number. Without making any changed during this time frame, last night it would just hang up but today I'm getting a busy signal. Just for grins and giggles I set up port forwarding on 5060 and 10001-20000, disabled fail2ban and stopped iptables, which I know shouldn't be necessary. It doesn't change anything and I don't imagine it will.

My gut tells me my side of the fence is OK. Asterisk is showing the SIP registration and Obvoice's customer portal is also showing the registration with the PBX server. It really sounds like a telco issue at their head end. I have emailed their support but at the slightest hint I wasn't using an Obihai device, I was dead to them.

Any other suggestions? If not and I'm at the end of my rope, I'll continue testing a couple of other providers though the deals aren't as great. In the interim I appreciate the guidance!
 

Todd Robinson

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And just to add to my gut feeling, I downloaded their Android app and still cannot receive calls. Same exact behavior when I dial, rings once and then busy signal. I contacted their support address and will update you all.
 

Todd Robinson

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Good grief...so buried in the Obivoice settings is a schedule in which calls can be automatically rejected. It was set to only allow calls from 12AM-12:05AM. Apparently that was the default setting out of the gate. Thanks for the suggestions and responses, guys!
 

wardmundy

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Please report this to the Obivoice folks. They'll want to fix the default setting. Glad you found the problem.
 

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