Re: [asterisk-users] hijacked thread

[Pete Mundy]

Sorry George. You're quite right, that was bad etiquette. I should have started 
a new thread with my reply to the hijack.

Pete

> On 22/03/2016, at 4:04 pm, George Joseph  wrote:
> 
> Now do you mind if we get back to the original purpose of this thread before 
> it was hijacked?
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Re: [asterisk-users] Loss of devices registration (pjsip)

[George Joseph]

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  wrote:

>
> Good result! Glad it worked for you :)
>
> Pete
>
>
> On 22/03/2016, at 9:34 am, somsad khan  wrote:
>
> 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 
> 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 
>> 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  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
>>>
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>>> 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)

[Pete Mundy]

Good result! Glad it worked for you :)

Pete


> On 22/03/2016, at 9:34 am, somsad khan  wrote:
> 
> 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  > 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  > 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 > > 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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Re: [asterisk-users] Loss of devices registration (pjsip)

[Pete Mundy]

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

> On 22/03/2016, at 9:28 am, somsad khan  wrote:
> 
> 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. 


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

[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 
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  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  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)

[George Joseph]

On Mon, Mar 21, 2016 at 11:58 AM, 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
>
​The client just registered​


> [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
>
​We added a new contact​


> [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
>
​We deleted the old contact​


> [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
>
​We qualified the contact successfully​


> [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
>
​At the next qualify, we couldn't reach the contact

[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
>
​The client just registered​

​(again)​

> [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
>
​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​


> [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
>
​We qualified the contact successfully​


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

[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  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  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)

[Richard Mudgett]

On Mon, Mar 21, 2016 at 2:49 PM, somsad khan 
wrote:

> 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
>

Please do not hijack threads.

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

[Pete Mundy]

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

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

[Dmitriy Serov]

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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