2 Devices Same *actual* Extension – Can It Be Done

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Hello,

My company has invested heavily in Counterpath’s Stretto provisioning platform for Mobile and Desktop VoIP clients .

At this time their system allows 2 devices (for example iPhone + desktop computer) using the same software license per user , which many of our users require.

Their provisioning system assumes that both devices will use the same SIP extension for auth however.

Normally we would use separate extensions and a follow-me , but if there is any way to use the same extension, I need to figure it out.

Thank you,

Kevin Long

7 thoughts on - 2 Devices Same *actual* Extension – Can It Be Done

  • Kevin Long wrote:

    PJSIP in Asterisk 13 can be configured to allow multiple registrations to a single AOR and with some changes to dialplan all are dialed when called.


    Joshua Colp Digium, Inc. | Senior Software Developer
    445 Jan Davis Drive NW – Huntsville, AL 35806 – US
    Check us out at: http://www.digium.com & http://www.asterisk.org

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    Hi,Le 09/03/2016 08:40, Kevin Long a

  • With Asterisk 13 you may be able to do it with PJSIP using two separate connections on the same AOR
    I believe you would have two separate endpoints that would register under the same user and auth. If I understand it correctly when you send a call to the AOR both registered endpoints would be rung. I have not tried inbound ring yet, but when I have registered for out bound multiple connections and it seems to work well.

    Bryant

    From: “Kevin Long”
    Sent: Wednesday, March 9, 2016 1:42 PM
    To: “Asterisk Users Mailing List – Non-Commercial Discussion”
    Subject: [asterisk-users] 2 devices same *actual* extension – can it be done

    Hello,

    My company has invested heavily in Counterpath’s Stretto provisioning platform for Mobile and Desktop VoIP clients .

    At this time their system allows 2 devices (for example iPhone + desktop computer) using the same software license per user , which many of our users require.

    Their provisioning system assumes that both devices will use the same SIP extension for auth however.

    Normally we would use separate extensions and a follow-me , but if there is any way to use the same extension, I need to figure it out.

    Thank you,

    Kevin Long

  • Can someone tell me if this is possible?

    I currently have a VOIP phone registered on an Asterisk PBX at a remote location (working fine). I want to install an Asterisk PBX at the local location. I will be porting the current POSTS lines to SIP trunking. So now I want the remote line and the local lines to appear on the same handset. This would mean I would have to pass internet to the phone for the remote extension and also register the local extensions. So, for example, I could have the remote extension assigned to line one
    (ACCOUNT 1 on the Polycom handset), and the local extensions assigned to lines two, three, and four ( ACCOUNTS 2,3,4).

    How do I do this?

    Thanks,

  • So, the first thing you will have to do is to make sure that your phone has routes to and can talk to each pbx over the network. Depending on your network design, this may be pretty simple or it may get pretty complex and will be hard to give a definitive answer in this discussion without more details. A good test might be to see if the phone can ping the pbx. Since you specifically mentioned a Polycom handset, look under Menu-Status-Diagnostics-Network-Ping. This will possibly help you to know that you can reach the pbx from the phone (provided your network is set up correctly and the pbx responds to pings). Note, many network designs will actually block pings even when the SIP and RTP traffic will traverse it just fine, so a failure here isn’t necessarily the kiss of death.

    Next, you will need to set up your phone to register with each PBX. Polycom has excellent docs on how to perform a setup using xml configuration files. Here is an example with four lines connecting to four different voip servers on a Polycom phone. Please note that I do not endorse the insecure usernames and passwords used here. They don’t follow best practices and are only here for an example.

    Note that this is just one small section out of a much larger configuration file used to completely configure a Polycom phone. Assuming you have the rest of your configs working, this would then put 4 lines onto the phone, each pointing to a different pbx and each labeled uniquely.

  • Hi!

    10.03.2016 20:12, asterisk-users-request@lists.digium.com пишет:

    And what would the behaviour be with , if a newly registered peer with same SIP login has appeared while the Dial is already progressing ?

    We’ve seen with chan_sip that there’s no straightforward manner to add a newly registered peer (think mobile application that registers after receiving push notification) to a progressing (not answered) call

    Thanks..

    Kirill

  • Kirill Marchuk wrote:

    The Asterisk dialing process itself does not allow this. Once channels are dialed you can’t add. You’d need to send the push notification, wait a period of time, and then do the Dial.