Re: [asterisk-users] Asterisk now available with bundled pjproject!

[Jean-Denis Girard]

Hi George,

It seems configure with --disable-pa, and configuration "#define
PJSIP_MAX_PKT_LEN 6000" did not make it to 13.8.0-rc1, do you still
intend to add include these modifications?


Thanks,
-- 
Jean-Denis Girard

SysNuxSystèmes   Linux   en   Polynésie   française
http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27

Le 13/03/2016 17:32, George Joseph a écrit :
> 
> 
> On Sat, Mar 12, 2016 at 10:48 PM, Jean-Denis Girard  > wrote:
> 
> Hi George,
> 
> Le 07/03/2016 12:53, George Joseph a écrit :
> > Le 07/03/2016 09:28, George Joseph a écrit :
> > > PLEASE TRY THIS!!  I'd love some feedback BEFORE 13.8.0 is 
> released.
> 
> I don't think this is related to the bundled version, but I got
> PJSIP_ERXOVERFLOW when initiating a WebRTC video call from Chrome:
> 
> [Mar 12 19:08:37] ERROR[9071]: pjproject:0 : sip_endpoint.c
> Error processing packet from 192.168.10.88:50072
> : Rx buffer overflow
> (PJSIP_ERXOVERFLOW)  [code 171062]:
> INVITE sip:*9...@sysnux.pf  SIP/2.0
> Via: SIP/2.0/WSS ca4cqpd5cv2h.invalid;branch=z9hG4bK2286368
> Max-Forwards: 70
> To: >
> From:  >;tag=q1ejnhm074
> Call-ID: l7rivm3clnebl6om63eb
> CSeq: 1487 INVITE
> Authorization: Digest algorithm=MD5, username="websip2",
> realm="asterisk", nonce="1457845717/bfbd52f55e31f89cda00a1305c272bd6",
> uri="sip:*9...@sysnux.pf ",
> response="d30a2f2b4d5d25e81dded44b7d98e336",
> opaque="639fdd14224f0290", qop=auth, cnonce="r0d44vjitbof", nc=0001
> Contact: 
> Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY
> Content-Type: application/sdp
> Supported: outbound
> User-Agent: SIP.js/0.7.3
> Content-Length: 3335
> ...
> 
> This can be solved by adding the following line to config_site.h:
> #define PJSIP_MAX_PKT_LEN   6000
> 
> Would you consider adding it?
> 
> 
> 
> Yes.  I'll add it this week.​
>  
> 
> 
> 
> Thanks,
> --
> Jean-Denis Girard
> 
> SysNuxSystèmes   Linux   en   Polynésie   française
> http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27
> 
> 




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[asterisk-users] Create conference bridge via outlook

[Ryan, Travis]

Is there any way to have a meeting request in Outlook allow someone to 
attach/setup a conference bridge, time, etc for Asterisk?


Travis Ryan
Director of Information Technologies
Oscar Winski Company
2407 North Ninth Street
Lafayette, IN 47905
ry...@oscarwinski.com
(765) 742-1102

We're not the IT departmentWe're the I-TEAM department!

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Re: [asterisk-users] Peer matching with PJSIP [SOLVED]

[Olivier]

2016-03-22 15:44 GMT+01:00 Joshua Colp :

> Olivier wrote:
>
>> Hello,
>>
>> I'm trying to understand how to configure Asterisk 13's PJSIP stack.
>> I've read the pages in [1] and still have a couple of questions to ask.
>>
>> In my lab, I've configured an Asterisk 13 box to act as an ITSP box and
>> another Asterisk 13 to act as an IPBX.
>>
>> I'm trying to configure the ITSP box such as it would accept incoming
>> registrations and calls from any IP address (dynamic), if and only if
>> IPBX provides both a username and a password.
>>
>> 1. From experiments, I've found that OPTION messages sent by IPBX box
>> have a From field which contains the following value :
>> From: > > >>;tag=cf192aac-7799-4d6d-be1a-8297125ee595
>> Is it possible to change this "asterisk" in the above From field ?
>> Aor section in [2] do not mention such setting.
>> I couldn't see any undesirable side effects (beside WARNING messages on
>> the ITSP box) from this.
>>
>
> This is controlled using from_user in the endpoint but does not currently
> work, there is a fix up[1] to improve this.
>
>
>> 2. From experiments, I've found that if INVITE messages from IPBX box do
>> not contain a From field with a callerid matching the peer section in
>> ITSP configuration, ITSP doesn't match incoming INVITE with appropriate
>> endpoint.
>> Is it possible and desirable to change ITSP box config so that any
>> INVITE coming from the same IP+port attached to previous REGISTER would
>> be matched appropriately ?
>>
>
> chan_pjsip does not currently support matching based on the IP
> address+port that a device has registered from. You can only explicitly
> configure this right now.
>
> [1] https://gerrit.asterisk.org/#/c/2373/
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>


Thank you very much for those detailed and exact answers.
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Re: [asterisk-users] WRONG Queues log

[Olivier]

Which Asterisk version do you use ?

And what is your exact question ?
What do you expect queue_log file to hold ?



2016-03-15 10:33 GMT+01:00 Антон Сацкий :

>
> Hi list need your help
> i have call in queue it shows that it was answered by 4003
> 
> [root@asterisk ~]# grep --color "1456128646.157422"
> /var/log/asterisk/queue_log-20160228
>
> 1456128688|1456128646.157422|800|NONE|ENTERQUEUE||0967145750|2
> 1456128717|1456128646.157422|800|SIP/4003|CONNECT|29|1456128688.157426|28
> 1456128817|1456128646.157422|800|SIP/4003|COMPLETECALLER|29|100|2
>
> 
> BUT IN FACT call was PICK UPPED  by 4001  using features
>
> [root@asterisk ~]# grep --color "1456128646.157422"
> /var/log/asterisk/full-20160228
> [Feb 22 10:11:28] VERBOSE[9760][C-f165] pbx.c: -- Executing
> [~~s~~@mix:2] MSet("SIP/3590640-000209b9",
> "CDR(recordingfile)=3590640_1456128646.157422") in new stack
> [Feb 22 10:11:28] VERBOSE[9760][C-f165] pbx.c: -- Executing
> [~~s~~@mix:3] MixMonitor("SIP/3590640-000209b9",
> "3590640_1456128646.157422.wav,b") in new stack
>
>
>
> [root@asterisk ~]# grep --color "C-f165"
> /var/log/asterisk/full-20160228
> [Feb 22 10:10:46] VERBOSE[2070][C-f165] netsock2.c:   == Using SIP RTP
> CoS mark 5
> [Feb 22 10:10:46] VERBOSE[9760][C-f165] pbx.c: -- Executing
> [3590640@incoming:1] Set("SIP/3590640-000209b9", "CALLERID(name)=RU") in
> new stack
> [Feb 22 10:10:46] VERBOSE[9760][C-f165] pbx.c: -- Executing
> [3590640@incoming:2] GotoIfTime("SIP/3590640-000209b9",
> "9:00-19:30,mon-fri,*,*?4") in new stack
> [Feb 22 10:10:46] VERBOSE[9760][C-f165] pbx.c: -- Goto
> (incoming,3590640,4)
> [Feb 22 10:10:46] VERBOSE[9760][C-f165] pbx.c: -- Executing
> [3590640@incoming:4] Goto("SIP/3590640-000209b9", "working") in new stack
> [Feb 22 10:10:46] VERBOSE[9760][C-f165] pbx.c: -- Goto
> (incoming,3590640,13)
> [Feb 22 10:10:46] VERBOSE[9760][C-f165] pbx.c: -- Executing
> [3590640@incoming:13] Progress("SIP/3590640-000209b9", "") in new stack
> [Feb 22 10:10:46] VERBOSE[9760][C-f165] pbx.c: -- Executing
> [3590640@incoming:14] MSet("SIP/3590640-000209b9", "EXT=3590640") in new
> stack
> [Feb 22 10:10:46] VERBOSE[9760][C-f165] pbx.c: -- Executing
> [3590640@incoming:15] Set("SIP/3590640-000209b9", "CHANNEL(language)=ru")
> in new stack
> [Feb 22 10:10:46] VERBOSE[9760][C-f165] pbx.c: -- Executing
> [3590640@incoming:16] Playback("SIP/3590640-000209b9",
> "01_HELLO/01_HELLO") in new stack
> [Feb 22 10:10:46] VERBOSE[9760][C-f165] res_rtp_asterisk.c:>
> 0x7f9b1c19d490 -- Probation passed - setting RTP source address to
> 95.67.3.3:14380
> [Feb 22 10:10:46] VERBOSE[9760][C-f165] file.c: --
>  Playing '01_HELLO/01_HELLO.slin' (language 'ru')
> [Feb 22 10:10:49] VERBOSE[9760][C-f165] pbx.c: -- Executing
> [3590640@incoming:17] Wait("SIP/3590640-000209b9", "2") in new stack
> [Feb 22 10:10:51] VERBOSE[9760][C-f165] pbx.c: -- Executing
> [3590640@incoming:18] BackGround("SIP/3590640-000209b9",
> "02_CHOICE_LANGUAGES/02_CHOICE_LANGUAGES") in new stack
> [Feb 22 10:10:51] VERBOSE[9760][C-f165] file.c: --
>  Playing
> '02_CHOICE_LANGUAGES/02_CHOICE_LANGUAGES.slin' (language 'ru')
> [Feb 22 10:10:55] DTMF[9760][C-f165] channel.c: DTMF begin '2'
> received on SIP/3590640-000209b9
> [Feb 22 10:10:55] DTMF[9760][C-f165] channel.c: DTMF begin ignored '2'
> on SIP/3590640-000209b9
> [Feb 22 10:10:55] DTMF[9760][C-f165] channel.c: DTMF end '2' received
> on SIP/3590640-000209b9, duration 260 ms
> [Feb 22 10:10:55] DTMF[9760][C-f165] channel.c: DTMF end passthrough
> '2' on SIP/3590640-000209b9
> [Feb 22 10:11:00] VERBOSE[9760][C-f165] pbx.c:   == CDR updated on
> SIP/3590640-000209b9
> [Feb 22 10:11:00] VERBOSE[9760][C-f165] pbx.c: -- Executing
> [2@incoming:1] Set("SIP/3590640-000209b9", "CHANNEL(language)=ua") in new
> stack
> [Feb 22 10:11:00] VERBOSE[9760][C-f165] pbx.c: -- Executing
> [2@incoming:2] Set("SIP/3590640-000209b9", "CALLERID(name)=UA") in new
> stack
> [Feb 22 10:11:00] VERBOSE[9760][C-f165] pbx.c: -- Executing
> [2@incoming:3] Goto("SIP/3590640-000209b9", "ua_start,3590640,1") in new
> stack
> [Feb 22 10:11:00] VERBOSE[9760][C-f165] pbx.c: -- Goto
> (ua_start,3590640,1)
> [Feb 22 10:11:00] VERBOSE[9760][C-f165] pbx.c: -- Executing
> [3590640@ua_start:1] Set("SIP/3590640-000209b9", "CHANNEL(language)=ua")
> in new stack
> [Feb 22 10:11:00] VERBOSE[9760][C-f165] pbx.c: -- Executing
> [3590640@ua_start:2] Set("SIP/3590640-000209b9", "TIMEOUT(digit)=3") in
> new stack
> [Feb 22 10:11:00] VERBOSE[9760][C-f165] func_timeout.c: -- Digit
> timeout set to 3.000
> [Feb 22 10:11:00] VERBOSE[9760][C-f165] pbx.c: -- Executing
> [3590640@ua_start:3] BackGround("SIP/3590640-000209b9",
> "01_QUALITY_OF_THE_SERVICE/01_QUALITY_OF_THE_SERVICE") 

Re: [asterisk-users] Peer matching with PJSIP

[Joshua Colp]

Olivier wrote:

Hello,

I'm trying to understand how to configure Asterisk 13's PJSIP stack.
I've read the pages in [1] and still have a couple of questions to ask.

In my lab, I've configured an Asterisk 13 box to act as an ITSP box and
another Asterisk 13 to act as an IPBX.

I'm trying to configure the ITSP box such as it would accept incoming
registrations and calls from any IP address (dynamic), if and only if
IPBX provides both a username and a password.

1. From experiments, I've found that OPTION messages sent by IPBX box
have a From field which contains the following value :
From: >;tag=cf192aac-7799-4d6d-be1a-8297125ee595
Is it possible to change this "asterisk" in the above From field ?
Aor section in [2] do not mention such setting.
I couldn't see any undesirable side effects (beside WARNING messages on
the ITSP box) from this.


This is controlled using from_user in the endpoint but does not 
currently work, there is a fix up[1] to improve this.




2. From experiments, I've found that if INVITE messages from IPBX box do
not contain a From field with a callerid matching the peer section in
ITSP configuration, ITSP doesn't match incoming INVITE with appropriate
endpoint.
Is it possible and desirable to change ITSP box config so that any
INVITE coming from the same IP+port attached to previous REGISTER would
be matched appropriately ?


chan_pjsip does not currently support matching based on the IP 
address+port that a device has registered from. You can only explicitly 
configure this right now.


[1] https://gerrit.asterisk.org/#/c/2373/

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org


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[asterisk-users] Peer matching with PJSIP

[Olivier]

Hello,

I'm trying to understand how to configure Asterisk 13's PJSIP stack.
I've read the pages in [1] and still have a couple of questions to ask.

In my lab, I've configured an Asterisk 13 box to act as an ITSP box and
another Asterisk 13 to act as an IPBX.

I'm trying to configure the ITSP box such as it would accept incoming
registrations and calls from any IP address (dynamic), if and only if IPBX
provides both a username and a password.

1. From experiments, I've found that OPTION messages sent by IPBX box have
a From field which contains the following value :
From: ;tag=cf192aac-7799-4d6d-be1a-8297125ee595
Is it possible to change this "asterisk" in the above From field ?
Aor section in [2] do not mention such setting.
I couldn't see any undesirable side effects (beside WARNING messages on the
ITSP box) from this.

2. From experiments, I've found that if INVITE messages from IPBX box do
not contain a From field with a callerid matching the peer section in ITSP
configuration, ITSP doesn't match incoming INVITE with appropriate endpoint.
Is it possible and desirable to change ITSP box config so that any INVITE
coming from the same IP+port attached to previous REGISTER would be matched
appropriately ?


Best regards

[1] https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip
[2])
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Configuration_res_pjsip
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Re: [asterisk-users] Loss of devices registration (pjsip)

[Dmitriy Serov]

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:



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] One phone, many names / Was: Loss of devices registration (pjsip)

[Ishfaq Malik]

On 22 March 2016 at 08:55, A J Stiles  wrote:

> On Monday 21 Mar 2016, 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.
>
> Dead easy!  Done this before, in a very similar situation  (agent has to
> answer with a different name, depending on the number the customer
> dialled).
>
> All you need to do -- as long as the phone you are using is modern enough
> to
> support it -- is have in your dialplan, before the Dial() instruction to
> the
> agent's phone, an instruction like
> Set(CALLERID(name)=something)
> where "something" depends on ${EXTEN}.
>
> For example, if the numbers for the virtual companies are 731615, 701289
> and
> 718182, and the extension to ring is 301, you might do
>
> [from_pstn]
> ; 731615 is company ABC
> exten => 731615,1,NoOp(Call to 731615)
> exten => 731615,n,Set(CALLERID(name)=Company ABC)
> exten => 731615,n,Dial(301)
> exten => 731615,n,HangUp()
>
> ; 701289 is company XYZ
> exten => 701289,1,NoOp(Call to 701289)
> exten => 701289,n,Set(CALLERID(name)=Company XYZ)
> exten => 701289,n,Dial(301)
> exten => 701289,n,HangUp()
>
> ; 718182 is company PQR
> exten => 718182,1,NoOp(Call to 718182)
> exten => 718182,n,Set(CALLERID(name)=Company PQR)
> exten => 718182,n,Dial(301)
> exten => 718182,n,HangUp()
>
>
> For the agent to be able to dial out presenting different caller ID
> numbers,
> use prefixes such as 16, 17, 18 to indicate dialling out as different
> companies;
> strip out the prefix using ${EXTEN:2} to recover the number by skipping two
> digits from the beginning, and Set(CALLERID(num)=) as appropriate.
>
>
>
>
You can also use the A option in the Dial application to play an audio file
to the callee before the channels are bridged.

https://wiki.asterisk.org/wiki/display/AST/Application_Dial



-- 

Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)161 660 2350
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
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Re: [asterisk-users] Loss of devices registration (pjsip)

[Ishfaq Malik]

On 21 March 2016 at 20:32, George Joseph 
wrote:

>
>
> 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.
>
>
>
>
>
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).




-- 

Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)161 660 2350
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
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Re: [asterisk-users] One phone, many names / Was: Loss of devices registration (pjsip)

[A J Stiles]

On Monday 21 Mar 2016, 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.

Dead easy!  Done this before, in a very similar situation  (agent has to 
answer with a different name, depending on the number the customer dialled).

All you need to do -- as long as the phone you are using is modern enough to 
support it -- is have in your dialplan, before the Dial() instruction to the 
agent's phone, an instruction like
Set(CALLERID(name)=something)
where "something" depends on ${EXTEN}.

For example, if the numbers for the virtual companies are 731615, 701289 and 
718182, and the extension to ring is 301, you might do

[from_pstn]
; 731615 is company ABC
exten => 731615,1,NoOp(Call to 731615)
exten => 731615,n,Set(CALLERID(name)=Company ABC)
exten => 731615,n,Dial(301)
exten => 731615,n,HangUp()

; 701289 is company XYZ
exten => 701289,1,NoOp(Call to 701289)
exten => 701289,n,Set(CALLERID(name)=Company XYZ)
exten => 701289,n,Dial(301)
exten => 701289,n,HangUp()

; 718182 is company PQR
exten => 718182,1,NoOp(Call to 718182)
exten => 718182,n,Set(CALLERID(name)=Company PQR)
exten => 718182,n,Dial(301)
exten => 718182,n,HangUp()


For the agent to be able to dial out presenting different caller ID numbers, 
use prefixes such as 16, 17, 18 to indicate dialling out as different 
companies; 
strip out the prefix using ${EXTEN:2} to recover the number by skipping two 
digits from the beginning, and Set(CALLERID(num)=) as appropriate.


-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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