Loss Of Devices Registration (pjsip)

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Asterisk Users 11 Comments

Good day.

Asterisk 13.7.2, res_pjsip. There is a problem of loss of registration of several devices. This happens not on all devices, but problem devices a lot. Below is the log of registration of a contact of one device.

Is suspect two things:
1. delete a contact after the contact is added. But, like, it’s a feature of code that may already be fixed.
2. deleting a contact much earlier than the 90 seconds specified during the registration

Would be grateful for any clues.

Dmitriy Serov.

expiration settings:
[common-aor](!)
type=aor qualify_frequency`
default_expiration0
maximum_expiration`0
minimum_expiration

11 thoughts on - Loss Of Devices Registration (pjsip)

  • Hello guys,

    I need some help.

    I have a client coming who wants to assign 5 different numbers to one virtual employee SIP phone at his desk or softphone (Zoiper).

    which I can assign for the incoming or outgoing both.

    but the problem is which I might not understanding enough, that,

    e.g. when line 1 calls the virtual employee will answer “hello this is xyz company how can I help you”

    when line 2 calls the virtual employee will answer “hello this is abc company how can I help you”

    So it is important the employee can recognize which line is calling as they cannot say the wrong company name by mistake!

    please let me know if there is any possible ways.

    currently I have my freeepbx server which I have installed in a VPS server. so all my ZOIPER extension is registered to the Freepbx server with IAX
    protocol. and I have another Asterisk server at my local office for using SIP phones. basically my both server are connected with IAX protocol as SIP
    port are blocked in my country.

    please help if it’s possible. thanks in advance

  • Many desk phones support multiple simultaneous SIP registrations. You could use BLF buttons for each SIP registration and the operator uses the LEDs as their queue as to which account is ringing. Alternatively the phone’s UI may be able to indicate which account is ringing without the need for BLFs.

    Another option is to re-write the CALLERID(num) or CALLERID(name) to indicate the inbound line (eg prepend a string or number).

    Hopefully that gives you some food for thought 🙂

    Pete

  • hello Pete Mundy,

    thanks alot for your idea and reply. but unfortunately none of our SIP
    phone have the facilities to use multiple line and UI.

    I can see incoming numbers on my softphone(Zoiper) when a incoming call hits. I liked your incoming caller ID customize idea.

    Is it possible to add company name with incoming numbers. so that company name or any signal will appear with incoming call numbers, will be easy to identify by employee that call is coming into which number.

    thank you

  • ​The client just registered​

    ​We added a new contact​

    ​We deleted the old contact​

    ​We qualified the contact successfully​

    ​At the next qualify, we couldn’t reach the contact

    [2016-03-21 20:41:06] VERBOSE[3827] res_pjsip_registrar.c: Added contact ‘
    ​The client just registered​

    ​(again)​

    ​We added a new contact​

    [2016-03-21 20:41:06] VERBOSE[28019] res_pjsip/pjsip_configuration.c:
    Contact 17367/sip:17367@46.39.229.18:37910 has been deleted
    ​We deleted the old contact​

    ​We qualified the contact successfully​

    ​At the next qualify, we couldn’t reach the contact

    ​This looks like a client that’s going to sleep or a firewall that’s timing out connections. Asterisk is only deleting the contact on the next successful register because it’s replacing it. You need to figure out why the qualify is failing and why the client keeps registering.

  • I have added CID name prefix on inbound route. and it works fine 🙂 now I
    can simply forward five incoming routes to one extension. and as far as I
    guess, if I add CID name prefix for every number. it should work 🙂 thanks alot 🙂

  • Now do you mind if we get back to the original purpose of this thread before it was hijacked?

    Dmitriy… See my response further back. 🙂

  • Check if the router or firewall has a UDP port timeout option and increase it by a lot (I usually up it to an hour).

  • Thanks, George Joseph!

    Now a lot clearer the reasons for this behavior. It turns out that in the case of devices there are two ways to understand that they are “alive”:
    1. Registration from device to server
    2. qualify from the server to the client And the second way does not seems superfluous. Since calling to the device this way will be used. if qualify doesn’t working, then this call will not take place.

    Most likely the problem is that the device is behind two NAT (from your ISP and your own router). Can you advise how to configure the client in this case? Is it necessary to use a stun (did not seem to help, and it only works in the case of RTP) or proxy?

    Thanks.

    21.03.2016 23:32, George Joseph пишет: