Multiple GV outgoing on Obi thru FreePBX/Asterisk
azrobert:
Are you forking the call to SP2 on the OBi110?
What (service provider?) is defined on the OBi110 SP2?
If you don't fork the call and just send the call to the OBi200, does it work?
Do you have separate OBiHai accounts for each OBi or just a single account for both?
For the OBiON that's failing, which OBi is the Gateway OBi?
Are you calling the OBi200 from OBiON?
How are you calling from OBiON?
hapollo:
Ok, after some further testing, I realized what was causing the issues and so many headaches for me!
Most of my testing after the installation of the Asterisk was using an Obion App on an Android Phone here to call into the remote Obi110 via Obitalk and call back out. The logic being that since the remote Obi110 is too far away, I'd call into it and call out as if I was actually there rather than have my parents test it.
So, as an experiment, I turned off the Wifi on my Android and used Cellular Data to call out using the Obion App, the 503 error went away and forked calls did go out SP2 and VG on the Obi200 but ended as soon connected.
All I wanted to do was use the Asterisk to replace the GV authentication and Obi firmware updates needed.
Is it safe to assume, if I didn't change anything aside from the GV authentication originally asked above, everything else should function as it did prior to Asterisk?
I guess the underlying question is how do I properly test calls going out of the remote Obi110 will work when I can't physically be there? Obviously, calling out on a Obion App to the Obi110 to call back out causes errors which I can assume are what would be considered calling loops? Otherwise, will have to use the Obis in the most basic way until I can be there during year end holiday gatherings.
hapollo:
Well Obitalk forking finally resolved!
Turns out VG2 was wrongly defined as SPx(AsteriskIP:Port#) matching SPx xUserAgentPort when it should've simply been 5060.
In the end, I guess, if someone could post some simply rules of when to use 5060, when to change to match SPx and when/if you do change ports where the corresponding changes should be made in Obi, or Asterisk vice versa could be helpful and solve 80% of issues people have with errors or audio issues.
But I learned alot in a short time. Thanks all.
NoelB:
Quote from: hapollo on July 28, 2016, 04:32:34 pm
In the end, I guess, if someone could post some simply rules of when to use 5060, when to change
If you ignore changes you may like to make for added security you probably never need to change default ports. Every sip pkt contains a source port (src) and a destination port (dst). Each sip server such as asterisk listens on a specific port normally 5060 so all pkts sent to a sip server must contain a destination port that the server is listening on. The pkts can be sent from any src port you like . Mostly this will default to 5060 but this src port is often changed either by your router (if the server is not on your lan ) on the public side to address conflict with another client sending pkts to the same IP:5060 or by a setting change you may make to thwart port scanners. If you are going to change make it a significant change outside the likely scan range of scanners eg 4xxxx rather than say 5070. Each of your sip clients can use the same port on the same lan. The obi however defaults to a different port for each spx but other atas with multiply itsps are forced to all use the same port. This is handled by the nat in your router and you dont even know about it unless you can see the nat sessions set up as each pkt enters the internet.
Asterisk is just another sip server but without a nat between client and server but also a server that is able to be configured. Asterisk by default will listen on 5060 and all the sip clients such as obi will send to 5060 by default . If however you change the asterisk listening port then you must make the same change to the DST port the obi sends msgs too. When a pkt is sent to any sip servers including asterisk the server keeps a record of received= and rport= which are the source IP:port of all pkts sent to it so they just respond to the same IP:port that the pkt came from and the nat at your end ( if relevant) will send it on to the correct client.With the obi on default asterisk will read the src port as 5060/1/2/3 and will know where the response must be sent even though there will only be one IP address.
hapollo:
Okay after digesting all the info above and using OBi with FreePBX on GV for nearly a week. Tweaking things I managed to get multiple SIP providers onto Asterisk similar to the SPx's on Obi. This includes Localphone and Onesuite which were set up on Obi for lowest cost call routing via Gateways and Digitmaps. Now they are on Asterisk/FreePBX simple as Trunks with only Outbound Routes.
The only left for me to do is try to make it as similar as possible without having to dial prefixes. In other words, it the past, I had set up call routing to certain countries based on int dial codes using custom digitmaps to route out one SIP provider via SPx or Gateway on another.
It tried what I think is the same method using Match Pattern with Asterisk extension as the CallerID parm and no prefixes, but calls don't go out that Trunk as expected.
Any suggestions on what to look for to make this transition as seamless as possible?
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