[asterisk-users] Loss of devices registration (pjsip)

somsad khan ctrlz.network at gmail.com
Mon Mar 21 14:49:33 CDT 2016


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

On Mon, Mar 21, 2016 at 11:58 PM, Dmitriy Serov <serov.d.p at gmail.com> wrote:

> 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=60
> default_expiration=120
> maximum_expiration=600
> minimum_expiration=90
>
> log:
> [2016-03-21 20:39:58] VERBOSE[30251] res_pjsip_registrar.c: Added contact '
> sip:17367 at 46.39.229.18:37910' to AOR '17367' with expiration of 90 seconds
> [2016-03-21 20:39:58] VERBOSE[28019] res_pjsip/pjsip_configuration.c:
> Contact 17367/sip:17367 at 46.39.229.18:37910 has been created
> [2016-03-21 20:39:58] VERBOSE[28019] res_pjsip/pjsip_configuration.c:
> Contact 17367/sip:17367 at 46.39.229.18:27143 has been deleted
> [2016-03-21 20:39:58] VERBOSE[28019] res_pjsip/pjsip_configuration.c:
> Contact 17367/sip:17367 at 46.39.229.18:37910 is now Reachable.  RTT: 41.882
> msec
> [2016-03-21 20:41:01] VERBOSE[28019] res_pjsip/pjsip_configuration.c:
> Contact 17367/sip:17367 at 46.39.229.18:37910 is now Unreachable.  RTT:
> 0.000 msec
> [2016-03-21 20:41:06] VERBOSE[3827] res_pjsip_registrar.c: Added contact '
> sip:17367 at 46.39.229.18:60105' to AOR '17367' with expiration of 90 seconds
> [2016-03-21 20:41:06] VERBOSE[28019] res_pjsip/pjsip_configuration.c:
> Contact 17367/sip:17367 at 46.39.229.18:60105 has been created
> [2016-03-21 20:41:06] VERBOSE[28019] res_pjsip/pjsip_configuration.c:
> Contact 17367/sip:17367 at 46.39.229.18:37910 has been deleted
> [2016-03-21 20:41:06] VERBOSE[28019] res_pjsip/pjsip_configuration.c:
> Contact 17367/sip:17367 at 46.39.229.18:60105 is now Reachable.  RTT: 44.031
> msec
> [2016-03-21 20:42:09] VERBOSE[28019] res_pjsip/pjsip_configuration.c:
> Contact 17367/sip:17367 at 46.39.229.18:60105 is now Unreachable.  RTT:
> 0.000 msec
> [2016-03-21 20:42:14] VERBOSE[3827] res_pjsip_registrar.c: Added contact '
> sip:17367 at 46.39.229.18:52836' to AOR '17367' with expiration of 90 seconds
> [2016-03-21 20:42:14] VERBOSE[28019] res_pjsip/pjsip_configuration.c:
> Contact 17367/sip:17367 at 46.39.229.18:52836 has been created
> [2016-03-21 20:42:14] VERBOSE[28019] res_pjsip/pjsip_configuration.c:
> Contact 17367/sip:17367 at 46.39.229.18:60105 has been deleted
> [2016-03-21 20:42:14] VERBOSE[28019] res_pjsip/pjsip_configuration.c:
> Contact 17367/sip:17367 at 46.39.229.18:52836 is now Reachable.  RTT: 40.032
> msec
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160322/ea1838a2/attachment.html>


More information about the asterisk-users mailing list