TIPS How to connect a trunk to getjive.com service

[Harry]

Hello All,

i have VoIP service with getjive.com using a polycom phone that connects to them directly. I would like to instead connect a trunk to them instead of the phone and connect the phone to the PBX instead.
Some more info:
I setup IncrediblePBX on a Rasberry PI and google voice as a Trunk to be able to send faxes.
Had some ip phones lying around and decided to connect them to the system. They work so good, i would like to create a trunk to my home VoIP provider. Problem is they do not offer SIP service, just VoIP (or whatever the proper terminology is) for IP phones. My problem is that i cannot figure out how to create the Trunk. Tried reading up on it, tried a few configs but nothing worked.
Google voice was so easy to setup and since it is similar in function as what i want to do, i figure there must be a way. Just can't figure it out on my own.
I don't want to switch providers and get a dedicated SIP trunk since getjive.com is what i also use for my business lines.
Can someone help?

 

[MGD4me]

According to their White Paper found on-line, they definitely DO use SIP technology for their VoIP service. I would be surprised if they had used something else, but why re-invent the wheel?

I suspect by reading their literature you probably received one of their pre-configured IP phones all ready to go, out of the box. If 'Yes', then GetJinve have loaded in the necessary credentials so that the phone in question can connect to their service. It then would have been a simple plug-n-play exercise on your part. They may have even 'locked down' the configuration so that you can not tinker with their settings, or even view what they are. If so, then you would need to contact GetJive directly to get the necessary credentials to add a trunk to your PBX, and also get the admin password for the phone, so that you can then configure the phone to register to your PBX, and NOT to GetJive any more.

Hope this helps...

 

[Harry]

I actually configured my own phones with their service, i have the necessary credentials to add a device. The problem is HOW to do it in the pbx.
The credentials i have are for a phone or softphone (like Bria) to connect to the service. How do i do that on the the PBX and make it a system trunk than can place and recieve calls?

For example, my home phone is connected to getjive direct. How do i use those credentials to create a trunk (what type, where on the connectivity menu) to connect the pbx to them? Then how do i configure one of the phones connected to the PBX to use that trunk to send and recieve calls.
The Google trunks have their own menu and was really easy.

This is all a bit new and confusing to me now. I tried reading up on it but was not able to get it working on my own.

Thanks for your help,

According to their White Paper found on-line, they definitely DO use SIP technology for their VoIP service. I would be surprised if they had used something else, but why re-invent the wheel?

I suspect by reading their literature you probably received one of their pre-configured IP phones all ready to go, out of the box. If 'Yes', then GetJinve have loaded in the necessary credentials so that the phone in question can connect to their service. It then would have been a simple plug-n-play exercise on your part. They may have even 'locked down' the configuration so that you can not tinker with their settings, or even view what they are. If so, then you would need to contact GetJive directly to get the necessary credentials to add a trunk to your PBX, and also get the admin password for the phone, so that you can then configure the phone to register to your PBX, and NOT to GetJive any more.

Hope this helps...

 

[Harry]

Some more info:

the credentials i have for the phones are: username, password, domain
when trying to setup a SIP or Custom Trunk there are way more entries to fill out. In the SIP Trunk i tried filling in just what i had but it did not work. ( I can see from the GetJive console if an extension (phone) is connected).

According to their White Paper found on-line, they definitely DO use SIP technology for their VoIP service. I would be surprised if they had used something else, but why re-invent the wheel?

I suspect by reading their literature you probably received one of their pre-configured IP phones all ready to go, out of the box. If 'Yes', then GetJinve have loaded in the necessary credentials so that the phone in question can connect to their service. It then would have been a simple plug-n-play exercise on your part. They may have even 'locked down' the configuration so that you can not tinker with their settings, or even view what they are. If so, then you would need to contact GetJive directly to get the necessary credentials to add a trunk to your PBX, and also get the admin password for the phone, so that you can then configure the phone to register to your PBX, and NOT to GetJive any more.

Hope this helps...

 

[Jake]

My experience with Jive is that it is the whole package, phones and a hosted PBX. They don't offer trunking as far as I understand. I'd be surprised if you can get it to work as a trunk on any PBX.

The customer I have that is on Jive has not had that great of an experience. I tried to tell them not to go that route but well when they didn't listen I just have to chuckle now....and tell them to call Jive

 

[rikishi]

Some more info:

the credentials i have for the phones are: username, password, domain
when trying to setup a SIP or Custom Trunk there are way more entries to fill out. In the SIP Trunk i tried filling in just what i had but it did not work. ( I can see from the GetJive console if an extension (phone) is connected).

I know nothing about Jive, but here are some "generally conservative" settings for your trunk. In these settings, replace 12345 with your actual username, 67890 with your password, and sip.example.com with your domain.

Under PEER Details:
username=12345
fromuser=12345
type=peer
secret=67890
insecure=port,invite
host=sip.example.com
fromdomain=sip.example.com
disallow=all
allow=ulaw
defaultexpirey=3600
canreinvite=no

USER Context: 12345

Under USER Details:
type=user
secret=67890
insecure=port,invite
defaultexpirey=3600
context=from-trunk
canreinvite=no

Register String: 12345:[email protected]/12345

Make sure that the number you send to the trunk is in the format Jive requires, e.g. 12125551212. Also, you need an Outbound Route to send calls to the Jive trunk, and a default Inbound Route to your extension, IVR, etc.

If it doesn't work, please report:
Does Jive show you as registered?
Does Asterisk show you as registered?
What goes wrong on outgoing calls (does called phone ring, what do you hear, what do they hear, what shows up in PIAF log, what shows up in Jive log)?
What goes wrong on incoming calls (as above)?

 

[Dave Gray]

Also, you'll need to whitelist the server address in the firewall (add-fqdn)

 

[Harry]

Thank you for your response.
Had some time to work on this today:

Registration seems to work:
Host dnsmgr Username Refresh State Reg.Time
reg.jiveip.net:5060 N xxxxxxxxxxxx 285 Registered Fri, 03 Apr 2015 16:15:32
1 SIP registrations.
Jive side is not clear if registered or not. They assigned the credentials to an extension that does not exist in the jive dial plan i have. I am talking with their support about this.
Edited the outbound route to use the Jive trunk instead of Google trunk.
I disabled the google trunk. Outbound calls get "all circuits are busy" message.
Inbound calls don't ring at all, default inbound route was there already from the google setup, going straight to an extension.

I did not put in all the SIP settings you mentioned. I had to add the outboundproxy setting (i recieved a new login from jive and that setting was there in addition to the domain setting). Tried it without the outbound proxy, but same results.

PEER:
host=reg.jiveip.net
outboundproxy=company.jive.rtcfront.net
username=12345
secret=6789
type=peer
USER:
outboundproxy=company.jive.rtcfront.net
secret=6789
type=user
context=from-trunk


Again, thanks for your help.

-H

I know nothing about Jive, but here are some "generally conservative" settings for your trunk. In these settings, replace 12345 with your actual username, 67890 with your password, and sip.example.com with your domain.

Under PEER Details:
username=12345
fromuser=12345
type=peer
secret=67890
insecure=port,invite
host=sip.example.com
fromdomain=sip.example.com
disallow=all
allow=ulaw
defaultexpirey=3600
canreinvite=no

USER Context: 12345

Under USER Details:
type=user
secret=67890
insecure=port,invite
defaultexpirey=3600
context=from-trunk
canreinvite=no

Register String: 12345:[email protected]/12345

Make sure that the number you send to the trunk is in the format Jive requires, e.g. 12125551212. Also, you need an Outbound Route to send calls to the Jive trunk, and a default Inbound Route to your extension, IVR, etc.

If it doesn't work, please report:
Does Jive show you as registered?
Does Asterisk show you as registered?
What goes wrong on outgoing calls (does called phone ring, what do you hear, what do they hear, what shows up in PIAF log, what shows up in Jive log)?
What goes wrong on incoming calls (as above)?

 

[Harry]

Are you referring to my network firewall or the PBX internal firewall?
Where would that be to configure?

My network firewall is already whitelisted the necessary IPs and ports are forwarded properly.

Also, you'll need to whitelist the server address in the firewall (add-fqdn)

 

[rikishi]

Try first without outboundproxy.

For outgoing, try adding to PEER:
fromuser=12345
and retest.

For incoming, try adding to USER:
insecure=port,invite
and retest.

If you still have trouble, debug tools include:
What you see on the Asterisk console.
What gets added to the 'full' log, typically at /var/log/asterisk/full
SIP error responses shown by the Asterisk command "sip set debug peer reg.jiveip.net" (without the quotes)

 

[Harry]

I got it working with Jive with "rikishi"'s help. THANK YOU very much.

here's the working config:

CONNECTIVITY > TRUNKS > ADD TRUNK
set trunk name
outbound caller id

in PEER Details add the following:
username=12345
type=peer
secret=6789
outboundproxy=companyname.jive.rtcfront.net
host=reg.jiveip.net
fromuser=12345
In USER Context add the extension number provided by Jive

in USER Details add the following:
type=user
secret=6789
outboundproxy=companyname.jive.rtcfront.net
context=from-trunk
insecure=port,invite
In Register String add the following assuming username is 12345 and pw is 6789
12345:6789@@reg.jiveip.net/12345
SAVE and APPLY changes
in Outbound Routes set the appropriate Trunk Sequence
in Inbound Route set the appropriate path to the IVR or Extension.
I now have both Google Voice Trunk and Jive.com extension working for both inbound and outbound calls.
Again, Big Thanks to user rikishi's help.