Re: [asterisk-users] Fwd: PJSIP subscribe

[George Joseph]

On Wed, Jun 8, 2016 at 8:48 AM, Annus Fictus  wrote:

> Hello,
>
> How can I know if is a BUG and report on Asterisk-Jira?
>
Asterisk is sending the correct "ep:busy" status but the clients appear to
not be using it.  I'm not sure what else we can do on the Asterisk side.


> Thank you
>
> Regards
>
>
>  Mensaje reenviado 
> Asunto: PJSIP subscribe
> Fecha: Mon, 6 Jun 2016 19:13:35 +0200
> De: Annus Fictus  
> Para: Asterisk Users Mailing List - Non-Commercial Discussion
>  
>
> Hello,
>
> I'm trying to use presence with PJSIP and  I have a "issue".
>
> I created correctly hint priorities like:
>
> exten => 1000,hint,PJSIP/1000
> exten => 1001,hint,PJSIP/1001
>
> Extension 1000 can subscribe extension 1001 y vice-versa. The problem is
> when the extension 1000 make or receive a call. In the softphone where
> the extension is present on buddy list, the extension appear go offline.
> When hang-up the extension return on-line.
>
> On the Asterisk console the command core show hints show the correct
> state for the extension.
>
> On my tests I used XLite, Bria and Jitsi.
>
> Any hint?
>
> Thank you
>
> Regards.
>
>
>
>
> --
> _
> -- 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
>



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

Re: [asterisk-users] pjsip: occasional sip_transactio Unable to register REGISTER transaction (key exists)

[Richard Mudgett]

On Wed, Jun 8, 2016 at 11:57 AM, Michael Maier  wrote:

> On 06/06/2016 at 04:40 PM Richard Mudgett wrote:
> > On Sun, Jun 5, 2016 at 3:48 AM, Michael Maier 
> wrote:
> >
> >> Hello!
> >>
> >> I occasionally can see warnings like these during *idle* times in
> >> asterisk log (asterisk 13.7.2):
> >>
> >> [2016-06-05 06:11:51] WARNING[27817] pjsip: sip_transactio Unable to
> >> register REGISTER transaction (key xists)
> >> [2016-06-05 06:11:51] WARNING[27817] pjsip: sip_transactio Unable to
> >> register REGISTER transaction (key exists)
>
> [...]
>
> > Those key exist messages are due to a race condition.  From what I've
> > seen in the code the messages seem to be benign.
>
> Please - could you tell, where you found them? Unfortunately, I wasn't
> able to find them!
>

Those messages are created inside the PJSIP library code.  Specifically in
the
sip_transaction.c file.  Asterisk is just passing the message on.

Richard
-- 
_
-- 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

Re: [asterisk-users] pjsip: occasional sip_transactio Unable to register REGISTER transaction (key exists)

[Michael Maier]

On 06/06/2016 at 04:40 PM Richard Mudgett wrote:
> On Sun, Jun 5, 2016 at 3:48 AM, Michael Maier  wrote:
> 
>> Hello!
>>
>> I occasionally can see warnings like these during *idle* times in
>> asterisk log (asterisk 13.7.2):
>>
>> [2016-06-05 06:11:51] WARNING[27817] pjsip: sip_transactio Unable to
>> register REGISTER transaction (key xists)
>> [2016-06-05 06:11:51] WARNING[27817] pjsip: sip_transactio Unable to
>> register REGISTER transaction (key exists)

[...]

> Those key exist messages are due to a race condition.  From what I've
> seen in the code the messages seem to be benign.

Please - could you tell, where you found them? Unfortunately, I wasn't
able to find them!


Thanks,
Michael

-- 
_
-- 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

[asterisk-users] Fwd: PJSIP subscribe

[Annus Fictus]

Hello,

How can I know if is a BUG and report on Asterisk-Jira?

Thank you

Regards


 Mensaje reenviado 
Asunto: PJSIP subscribe
Fecha:  Mon, 6 Jun 2016 19:13:35 +0200
De: Annus Fictus 
Para: 	Asterisk Users Mailing List - Non-Commercial Discussion 





Hello,

I'm trying to use presence with PJSIP and  I have a "issue".

I created correctly hint priorities like:

exten => 1000,hint,PJSIP/1000
exten => 1001,hint,PJSIP/1001

Extension 1000 can subscribe extension 1001 y vice-versa. The problem is
when the extension 1000 make or receive a call. In the softphone where
the extension is present on buddy list, the extension appear go offline.
When hang-up the extension return on-line.

On the Asterisk console the command core show hints show the correct
state for the extension.

On my tests I used XLite, Bria and Jitsi.

Any hint?

Thank you

Regards.


-- 
_
-- 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

[asterisk-users] asterisk and freeradius AAA

[Willy Offermans]

Dear asterisk friends,

I like to use asterisk and to do authentication, authorization and 
accounting (AAA) for it with freeradius. I looked for any documentation on 
the net, but could not find much useful and detailed information. 

I have made a first shot with radiusclient-ng. I configured cdr.conf, 
cel.conf and radiusclient.conf. Execution of radexample gave a positive 
result. So at least for authentication, the setup of radiusclient.conf 
seems to work. However asterisk throws following error upon a phonecall:

[Jun  8 12:17:29] ERROR[101041]: cdr_radius.c:228 radius_log: Failed to record 
Radius CDR record!

The debug of freeradius does not show any activity.

Is it possible to combine asterisk and freeradius for AAA? If yes, is there 
any documentation available and can you point me to it? Can I debug the 
error message ERROR[101041] in more detail.

-- 
Met vriendelijke groeten,
With kind regards,
Mit freundlichen Gruessen,

Will

*
 W.K. Offermans

   Powered by 

(__)
 \\\'',)
   \/  \ ^
   .\._/_)

   www.FreeBSD.org

-- 
_
-- 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

Re: [asterisk-users] Delay after Answer

[Israel Gottlieb]

Another thing i would check is encryption is disabled on the snom
בתאריך 8 ביוני 2016 10:07,‏ "Israel Gottlieb"  כתב:

> Are you using stun? I have seen that when using stun
> בתאריך 8 ביוני 2016 09:54,‏ "Faheem Muhammad"  כתב:
>
>>
>>
>> Are you sure *nslookup  *command is returning as expected?
>> Also check the output of the below command.
>> >> hostname && hostname -s && hostname -f
>>
>>
>> On Tue, Jun 7, 2016 at 11:54 PM, Brent Davidson <
>> br...@texascountrytitle.com> wrote:
>>
>>> Well, I thought I had the problem solved.  Ported everything over to
>>> PJSip and build RDNS records for the phones and the server, but I am still
>>> experiencing the problem on incoming calls.
>>>
>>>
>>> On 6/7/2016 1:00 PM, Faheem Muhammad wrote:
>>>
>>> I've faced the same issue. The issue was related to DNS, the reverse
>>> lookup query failure caused the delay around(7-9 seconds). The purpose of
>>> reverse lookup is to block IP Spoofing attacks.
>>>
>>> Regards,
>>> Faheem
>>>
>>> On Tue, Jun 7, 2016 at 7:48 PM, Brent Davidson <
>>> br...@texascountrytitle.com> wrote:
>>>
 I am having an issue with a couple of phones where they ring, but there
 is a long delay after the phone is picked up before the audio starts.

 My setup:

- Server running Asterisk 13.9.1, Dahdi 2.11.1 w/ OSLEC
- Server is CentOS 7
- Quad core CPU with 16GB Ram
- 2 Snom 300 phones.
- NO NAT.  Server and phone are on the same subnet with only a
gigabit switch between them.
- Digium TDM400 analog card with 2 incoming analog PSTN lines

 When a call comes in, the system answers, IVR plays, caller dials an
 extension, Snom 300 rings, handset picked up.  Caller continues to hear
 ringing for another 7 to 10 seconds.  Answerer hears a click, a quick burst
 of audio, then silence, then another click and audio is engaged.

 I have tried both SIP and RTP debugging and there are absolutely no
 messages indicating any timeout or retransmit.  I am at a total loss.  In
 the past I've always been able to find an answer to issues like this on my
 own, but this time I just don't know.  I was even beginning to suspect the
 network switch might be bad, but pinging between the server and the phones
 shows no packet loss and 0.969ms average response time.

 What am I missing*?*
 Thanks,
 Brent Davidson

 --
 _
 -- 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

>>>
>>>
>>>
>>> --
>>> _
>>> -- 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
>>>
>>
>>
>> --
>> _
>> -- 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
>>
>
-- 
_
-- 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

Re: [asterisk-users] Delay after Answer

[Israel Gottlieb]

Are you using stun? I have seen that when using stun
בתאריך 8 ביוני 2016 09:54,‏ "Faheem Muhammad"  כתב:

>
>
> Are you sure *nslookup  *command is returning as expected?
> Also check the output of the below command.
> >> hostname && hostname -s && hostname -f
>
>
> On Tue, Jun 7, 2016 at 11:54 PM, Brent Davidson <
> br...@texascountrytitle.com> wrote:
>
>> Well, I thought I had the problem solved.  Ported everything over to
>> PJSip and build RDNS records for the phones and the server, but I am still
>> experiencing the problem on incoming calls.
>>
>>
>> On 6/7/2016 1:00 PM, Faheem Muhammad wrote:
>>
>> I've faced the same issue. The issue was related to DNS, the reverse
>> lookup query failure caused the delay around(7-9 seconds). The purpose of
>> reverse lookup is to block IP Spoofing attacks.
>>
>> Regards,
>> Faheem
>>
>> On Tue, Jun 7, 2016 at 7:48 PM, Brent Davidson <
>> br...@texascountrytitle.com> wrote:
>>
>>> I am having an issue with a couple of phones where they ring, but there
>>> is a long delay after the phone is picked up before the audio starts.
>>>
>>> My setup:
>>>
>>>- Server running Asterisk 13.9.1, Dahdi 2.11.1 w/ OSLEC
>>>- Server is CentOS 7
>>>- Quad core CPU with 16GB Ram
>>>- 2 Snom 300 phones.
>>>- NO NAT.  Server and phone are on the same subnet with only a
>>>gigabit switch between them.
>>>- Digium TDM400 analog card with 2 incoming analog PSTN lines
>>>
>>> When a call comes in, the system answers, IVR plays, caller dials an
>>> extension, Snom 300 rings, handset picked up.  Caller continues to hear
>>> ringing for another 7 to 10 seconds.  Answerer hears a click, a quick burst
>>> of audio, then silence, then another click and audio is engaged.
>>>
>>> I have tried both SIP and RTP debugging and there are absolutely no
>>> messages indicating any timeout or retransmit.  I am at a total loss.  In
>>> the past I've always been able to find an answer to issues like this on my
>>> own, but this time I just don't know.  I was even beginning to suspect the
>>> network switch might be bad, but pinging between the server and the phones
>>> shows no packet loss and 0.969ms average response time.
>>>
>>> What am I missing*?*
>>> Thanks,
>>> Brent Davidson
>>>
>>> --
>>> _
>>> -- 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
>>>
>>
>>
>>
>> --
>> _
>> -- 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
>>
>
>
> --
> _
> -- 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
>
-- 
_
-- 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

Re: [asterisk-users] Delay after Answer

[Faheem Muhammad]

Are you sure *nslookup  *command is returning as expected?
Also check the output of the below command.
>> hostname && hostname -s && hostname -f


On Tue, Jun 7, 2016 at 11:54 PM, Brent Davidson  wrote:

> Well, I thought I had the problem solved.  Ported everything over to PJSip
> and build RDNS records for the phones and the server, but I am still
> experiencing the problem on incoming calls.
>
>
> On 6/7/2016 1:00 PM, Faheem Muhammad wrote:
>
> I've faced the same issue. The issue was related to DNS, the reverse
> lookup query failure caused the delay around(7-9 seconds). The purpose of
> reverse lookup is to block IP Spoofing attacks.
>
> Regards,
> Faheem
>
> On Tue, Jun 7, 2016 at 7:48 PM, Brent Davidson <
> br...@texascountrytitle.com> wrote:
>
>> I am having an issue with a couple of phones where they ring, but there
>> is a long delay after the phone is picked up before the audio starts.
>>
>> My setup:
>>
>>- Server running Asterisk 13.9.1, Dahdi 2.11.1 w/ OSLEC
>>- Server is CentOS 7
>>- Quad core CPU with 16GB Ram
>>- 2 Snom 300 phones.
>>- NO NAT.  Server and phone are on the same subnet with only a
>>gigabit switch between them.
>>- Digium TDM400 analog card with 2 incoming analog PSTN lines
>>
>> When a call comes in, the system answers, IVR plays, caller dials an
>> extension, Snom 300 rings, handset picked up.  Caller continues to hear
>> ringing for another 7 to 10 seconds.  Answerer hears a click, a quick burst
>> of audio, then silence, then another click and audio is engaged.
>>
>> I have tried both SIP and RTP debugging and there are absolutely no
>> messages indicating any timeout or retransmit.  I am at a total loss.  In
>> the past I've always been able to find an answer to issues like this on my
>> own, but this time I just don't know.  I was even beginning to suspect the
>> network switch might be bad, but pinging between the server and the phones
>> shows no packet loss and 0.969ms average response time.
>>
>> What am I missing*?*
>> Thanks,
>> Brent Davidson
>>
>> --
>> _
>> -- 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
>>
>
>
>
> --
> _
> -- 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
>
-- 
_
-- 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