Cell Phone as extentions

[bpados]

I'm trying to migrate to PIAF from other systems so I installed a test box with PIAF 1.4 and asterisk 1.6 on and upgraded freepbx to 2.6, but something stops me from completing the migration. When I set up follow me's or transfer a call to a cell phone that set up as custom extension the phones ring and as soon as the phone is answered the call gets disconnected.
All other systems have the same specs as my test box and this feature works very well.

Anyone can give me a direction as to why this doesnt work?

Thanks.

 

[kenn10]

I don't know what is going on. I use cell phones in follow-me and ring groups with no issues. I do use the "Confirm" option on follow me and ring groups for calls to cell phones to insure phones turned off or busy don't go directly to the cell phone's voice mail.

A little more info is needed to trouble shoot however. What kind of trunking are the calls to the cell phones using, how your are setting up your follow-me or ring-groups, etc. Also, how are your cell phones provisioned? Do they have voice mail or just deny the call if they are turned off or busy?

 

[bpados]

All sip trunks. Two to be exact from two separate providers.
I use custom extensions and under this device uses custom technology I use dial: sip/trunkname/11digit cell number. No voicemail.
With follow me I simply use the extension # and have "confirm" checked.
What I’m hoping to achieve is to have a asterisk 1.6 based system that can accommodate skype channels eventually.

 

[etech3]

Have you tested dialing the cell phone directly?
How about transfer a call directly to the cell phone?

 

[bpados]

And as it turns out i'm not able to make any calls at all. I guess i had the test box up and running long enough to run a test to a few cell phones and I didnt even try hitting the phone # directly. So the same thing happens to all phone # not just custom extensions.
The calls stop at the following lines. As for the phone the call is made from it looks like the call was answered it's only on the cell phone i see the calls are disconnected already.

SIP/bv-00000003 is making progress passing it to SIP/202-00000002
-- SIP/bv-00000003 answered SIP/202-00000002
-- Packet2Packet bridging SIP/202-00000002 and SIP/bv-00000003

 

[etech3]

"as it turns out i'm not able to make any calls at all"

Is the softphone registered on freepbx?
what happens when you dial *60 on the softphone

 

[bpados]

that gives me the time at the sound of the tone.

 

[etech3]

OK good that means that the softphone is registered on the freepbx.


have you got more than one sip extension setup? can you call to another extension?
On the freepbx admin page, Look at the trunks, are they registered?
Have you got a outbound route set up?
are your trunks registered?
can you make a call?

 

[bpados]

there are 11 extensions and I can call back and forth. I can also call in. Have an outgoing route and its the exact replica of the previous TB box. No outgoing calls they all stop at.

SIP/bv-00000003 is making progress passing it to SIP/202-00000002
-- SIP/bv-00000003 answered SIP/202-00000002
-- Packet2Packet bridging SIP/202-00000002 and SIP/bv-00000003

 

[etech3]

Look on the admin page of freepbx, are the SIP trunks registered?

FreePBX Statistics

Total active calls 0

Internal calls 0

External calls 0

Total active channels 0

FreePBX Connections

IP Phones Online 2

IP Trunks Online 1

IP Trunk Registrations 1

 

[kenn10]

Also post your setting from the trunk screen and the name of your provider.

 

[bpados]

9 phones online
3 trunks online
2 trunk registration

 

[bpados]

Broadvoice.
max channels: 4
dial rules:
NXXNXXXXXX
NXXXXXX
1NXXNXXXXXX

Trunk name: bv

Peer Details:
username=<10 digit phone#>
user=phone
type=peer
secret=<my password>
nat=yes
insecure=very
host=sip.broadvoice.com
fromuser=<10 digit phone#>
fromdomain=sip.broadvoice.com
dtmfmode=rfc2833
dtmf=rfc2833
canreinvite=no
authname=<10 digit phone#>

User Context: proxy.dca.broadvoice.com

User Details:
username=<10 digit phone#>
user=phone
type=user
secret=<my password>
insecure=very
host=sip.broadvoice.com
fromuser=<10 digit phone#>
fromdomain=sip.broadvoice.com
dtmfmode=rfc2833
dtmf=rfc2833
context=from-trunk
canreinvite=no
authname=<10 digit phone#>

Register string: <phone#>@sip.broadvoice.com:<mypassword>:<phone#>@sip.broadvoice.com

 

[kenn10]

Your register string looks suspect since there should be no space after the @ sign.

See this link on the Broadvoice site: http://broadvoice.com/support_install_asterisk.html

Sip.conf

• Pedantic
  In the [general] section ensure pedantic is set to no
  pedantic=no
• Registration
  In the [general] section of the config file create a line like this:
  register => <phonenumber>@sip.broadvoice.com:<password>:<phonenumber>@sip.broadvoice.com/<extension>

Also, their recommended trunk settings look a little different from yours (type and user settings, dtmf and dtmfmode, ad I think user-context should be "from-broadvoice):

[sip.broadvoice.com]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=<phone number>
secret=<register password>
username=<phone number>
insecure=very
context=from-broadvoice
authname=<phone number>
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no

• Replace phonenumber with your account phone number,
• Replace password with your password
• Replace extension with one of your accessible extensions in the dial plan.

Finally, from the Asterisk CLI, do a "sip show registry" and see if you see the Broadvoice trunk registered. If not, you have trunk setup issues.

 

[jmullinix]

I know that registration string looks wrong but, if memory serves, that is the way Broadvoice does it. I stopped using Broadvoice years ago due to reliability problems.

 

[bpados]

Sorry guys but that typo in the registration string is only my copy and paste typo. The server registers fine and I can call in so inbound routes work and when i call out the remote phone such as cell phone rings but as soon as i answer they call disconnects.
Do you think that disconnect is part of an incorrect registration?

 

[bpados]

Also I changed my trunk settings to what one of the replies told me to. I'm still having the same issue. I can call in and get to extensions, I can call around in the office and call remote extensions, but when I call any outside # I get to ringing and as soon as the call is answered the remote end of the call gets disconnected and on the extension end it looks like someone answered the phone with complete silence.

 

[etech3]

How about setting up your registration to your provider on a soft phone and making a few calls to see what you get.

I suspect something in your registration may need a tweak.

Do the soft phone set up first and you may have to talk to your provider's tech support.

Post back.

 

[bpados]

I have two virutal systems one is a Trixbox and this new Pbxinaflash and what i've been doing is switching between the two back and forth. The trixbox system's been using this registration for nearly 4 years now no problem. That's why i'm pausled about this issue.

 

[etech3]

run the tb asterisk CLI and capture when you dial a successfull call and do the same for the PIAF. Compare the two and see what you get.