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[serov.d.p at gmail.com]

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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[ctrlz.network at gmail...]

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@gmail.com (serov.d.p@gmail.com)> wrote:

[pete at fiberphone.co.nz]

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

[rmudgett at digium.com]

On Mon, Mar 21, 2016 at 2:49 PM, somsad khan <ctrlz.network@gmail.com (ctrlz.network@gmail.com)> wrote:

[ctrlz.network at gmail...]

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 <pete@fiberphone.co.nz (pete@fiberphone.co.nz)> wrote:

[george.joseph at fairv...]

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

[ctrlz.network at gmail...]

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.network@gmail.com (ctrlz.network@gmail.com)> wrote:

[pete at fiberphone.co.nz]

Somsad,

Yep. That's why I suggested it as another option

These links may help:

http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List
(see CALLERID(num) and CALLERID(name))

http://www.voip-info.org/wiki/view/Asterisk+cmd+Set

Pete

[pete at fiberphone.co.nz]

Good result! Glad it worked for you
Pete

[george.joseph at fairv...]

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.




On Mon, Mar 21, 2016 at 8:42 PM, Pete Mundy <pete@fiberphone.co.nz (pete@fiberphone.co.nz)> wrote:

[ish at pack-net.co.uk]

On 21 March 2016 at 20:32, George Joseph <george.joseph@fairview5.com (george.joseph@fairview5.com)> wrote:

[serov.d.p at gmail.com]

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: