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

2016-03-21 Thread somsad khan 
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  :)

On Tue, Mar 22, 2016 at 2:28 AM, somsad khan <ctrlz.netw...@gmail.com>
wrote:

> 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
>
> On Tue, Mar 22, 2016 at 2:12 AM, Pete Mundy <p...@fiberphone.co.nz> wrote:
>
>>
>> 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
>>
>>
>> On 22/03/2016, at 8:49 am, somsad khan <ctrlz.netw...@gmail.com> wrote:
>> 
>>
>> I have a client coming who wants to assign 5 different numbers to one
>> virtual employee SIP phone at his desk or softphone (Zoiper).
>>
>> 
>>
>> please let me know if there is any possible ways.
>>
>>
>> --
>> _
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>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Re: [asterisk-users] Loss of devices registration (pjsip)

2016-03-21 Thread somsad khan 
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

On Tue, Mar 22, 2016 at 2:12 AM, Pete Mundy <p...@fiberphone.co.nz> wrote:

>
> 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
>
>
> On 22/03/2016, at 8:49 am, somsad khan <ctrlz.netw...@gmail.com> wrote:
> 
>
> I have a client coming who wants to assign 5 different numbers to one
> virtual employee SIP phone at his desk or softphone (Zoiper).
>
> 
>
> please let me know if there is any possible ways.
>
>
> --
> _
> -- 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
>
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Re: [asterisk-users] Loss of devices registration (pjsip)

2016-03-21 Thread somsad khan 
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  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@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@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@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@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@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@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@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@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@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@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@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@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@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@46.39.229.18:52836 is now Reachable.  RTT: 40.032
> msec
>
>
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