Asterisk and Planet VGW-400FO

[aribic]

Hi, some beginner problems here!
I’m having some problems pairing Planet VGW-400FO (FXO) with Asterisk. Situation is as follows:
Asterisk up and running, IP phones set up and internal communication working as should.
Managed to create sip trunk between the gw and asterisk, when calling to pstn line phone rings (hot line extension) not showing caller ID, when I answer the call phone goes to please hang up while on the caller side is still ringing.
on the outbound side when trying to call to the outside trough trunk I receive all lines are busy message.
really stuck on this for quite a while, any help appreciated.

[david551]

Please provide the detailed protocol traces from chan_sip or chan_pjsip.

[aribic]

Hi @david551 tnx for answering.sorry but how do I do that?

[aribic]

here are the config screenshots

[aribic]

[aribic]

[aribic]

[david551]

https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information#CollectingDebugInformation-Enablechanneltechorfeaturespecificdebug

Please note this forum doesn’t support GUI front ends to Asterisk and Elastix will not support the hardware you are using.

[aribic]

So basically I should drop everything and install plain Asterisk to get it to work with Planet FXO?

[david551]

Not necessarily, but you are on your own in debugging the problem, unless you can present the underlying Asterisk problem, and you are on your own in terms of translating the result back into an Elastix configuration.

If you provide the debugging information, we may still be able to say what is wrong, but it will be in terms of the contents of sip.conf and extensions.conf, not in terms of Elastix GUI web page contents.

[aribic]

Sorry for the delayed response, I was unable to get to the office during weekend.
Since it’s not allowing me to upload files (new user) sip log is on the following link.
https://www.dropbox.com/s/d5cacx7j9v0lfip/log1.txt?dl=0
For easier getting around:
192.168.1.240 - asterisk / elastix, .241 planet FXO, rest of .24x phones.

thanks for all the help.

[david551]

That log shows a successful call that was cleared by 4PORT_GW. There are no timestamps (you probably screen scraped, rather than using the log files), so I can’t tell if the gateway dropped the call immediately after the ACK

There is no line saying that Asterisk is seeing a busy condition.

[aribic]

sorry, was a bit confused the first time. Did is as supposed this time.
So, after responding the on the call phone gives “pls. hang up” while on still ringing on the caller side. after a few seconds it restarts ringing.

Did an outgoing call try after the incoming one.

https://www.dropbox.com/s/mxyahx6bv500xry/debug_log_community1.txt?dl=0

[david551]

You need the full log, as the protocol specific traces are output as verbose, not debug, messages.

[david551]

There is really too much noise in the log, but it appears that 192.168.1.241:5060 initiated clearing the call and Asterisk tried to clear the call to 192.168.1.241:5060 as a result. However, there seems to be more than one call in the log.

[aribic]

Could be since after I answer the phone phone gives pls. hang up (on the caller side still ringing) and after few seconds phone rings again.