[asterisk-users] Double queue calls being delivered to agents

[Derek Bolichowski]

I posted this over in asterisk-dev, realized I probably should have put it 
here. 

Hi there,
We’ve been having a strange issue with a customer’s queues where a queued call 
will ring an available agent, agent answers, then a second or two later the 
agent is offered a second call which they cannot answer, since they’re already 
speaking with a client.

This in turn causes a few issues:
- Agent stats are no longer accurate, as it gets marked down as a ‘missed call’.
- Cannot use ‘autopause’ feature any longer, as the second queue call goes 
unanswered and pauses the agents.

The basic queue setup is as follows:
Autofill = yes
Ringinuse = no
Wrapuptime = 5
Strategy = fewestcalls (tried ‘random’ also)
Timeout = 15

We’re on Asterisk 11.21.2 currently.

In talking to a few colleagues, they seem to recall there being an old patch 
for the Asterisk queues application that inserted a short 100ms delay between 
delivering first and second calls.  I’ve scoured the web today, and found some 
old forums posts of people looking for something exactly like this, but haven’t 
found the actual patch, if one even exists.

I’m hoping someone may have some suggestions on some options we can try to 
eliminate this issue.

Thanks for taking the time to read this.

-Derek B
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[asterisk-users] Double queue calls being delivered to agents

[Derek Bolichowski]

Sorry for last post -- forgot to wipe out the digest contents :/ 

Derek B

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Re: [asterisk-users] Double queue calls being delivered to agents

[Derek Bolichowski]

Awesome. Thanks again Richard.


On May 4, 2016, at 10:59 PM, Richard Mudgett 
<rmudg...@digium.com<mailto:rmudg...@digium.com>> wrote:



On Tue, May 3, 2016 at 8:59 PM, Richard Mudgett 
<rmudg...@digium.com<mailto:rmudg...@digium.com>> wrote:


On Tue, May 3, 2016 at 6:15 PM, Derek Bolichowski 
<de...@empire-team.com<mailto:de...@empire-team.com>> wrote:
I posted this over in asterisk-dev, realized I probably should have put it here.

Hi there,
We've been having a strange issue with a customer's queues where a queued call 
will ring an available agent, agent answers, then a second or two later the 
agent is offered a second call which they cannot answer, since they're already 
speaking with a client.

This in turn causes a few issues:
- Agent stats are no longer accurate, as it gets marked down as a 'missed call'.
- Cannot use 'autopause' feature any longer, as the second queue call goes 
unanswered and pauses the agents.

The basic queue setup is as follows:
Autofill = yes
Ringinuse = no
Wrapuptime = 5
Strategy = fewestcalls (tried 'random' also)
Timeout = 15

We're on Asterisk 11.21.2 currently.

In talking to a few colleagues, they seem to recall there being an old patch 
for the Asterisk queues application that inserted a short 100ms delay between 
delivering first and second calls.  I've scoured the web today, and found some 
old forums posts of people looking for something exactly like this, but haven't 
found the actual patch, if one even exists.

I'm hoping someone may have some suggestions on some options we can try to 
eliminate this issue.

Thanks for taking the time to read this.

This issue has been around a long time and was just recently fixed and I think
it was just released in the latest v11 version.
See https://issues.asterisk.org/jira/browse/ASTERISK-16115

Looks like it will be in the next release as the issue does not have a target 
release set.

Richard

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[asterisk-users] Double queue calls being delivered to agents

[Derek Bolichowski]

I took a look through Asterisk 11 and 13 change logs but didn't see any mention 
of that patch/fix. Am I missing something?

Derek B

> On May 4, 2016, at 8:50 AM, "asterisk-users-requ...@lists.digium.com" 
> <asterisk-users-requ...@lists.digium.com> wrote:
> 
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> 
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-users digest..."
> 
> 
> Today's Topics:
> 
>   1. Re: Asterisk 13 Realtime Voicemail frustratingissue
>  (John Kiniston)
>   2. Re: Migrating asterisk 11 to 13: some callers get no ringback
>  tone any more (Michael Maier)
>   3. Re: Migrating asterisk 11 to 13: some callers get no ringback
>  tone any more (Joshua Colp)
>   4. Re: Migrating asterisk 11 to 13: some callers get no ringback
>  tone any more (Eric Wieling)
>   5. Re: Migrating asterisk 11 to 13: some callers get no ringback
>  tone any more (Joshua Colp)
>   6. Call a subroutine via Originate? (John Kiniston)
>   7. Re: Call a subroutine via Originate? (Bruce Ferrell)
>   8. Double queue calls being delivered to agents (Derek Bolichowski)
>   9. Execute an app on the master channel from inside a Macro on
>  the called channel (Saint Michael)
>  10. Re: Double queue calls being delivered to agents (Richard Mudgett)
>  11. Re: Migrating asterisk 11 to 13: some callers get no ringback
>  tone any more (Michael Maier)
>  12. Re: T.38 with Audiocodes gateway [SOLVED] (Olivier)
>  13. Asterisk registers with TLS,but sends out calls via UDP
>  (Sebastian Damm)
>  14. Compatibilty between agi for asterisk 13.8.0 andphp5.6
>  (Mamadou NGOM)
> 
> 
> --
> 
> Message: 1
> Date: Tue, 3 May 2016 11:39:44 -0700
> From: John Kiniston <johnkinis...@gmail.com>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
><asterisk-users@lists.digium.com>
> Subject: Re: [asterisk-users] Asterisk 13 Realtime Voicemail
>frustratingissue
> Message-ID:
><cafjqogc8syl_fsl8pmr+p6f6p1-nzk-_3rayrakw4kzjev8...@mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
> 
> Have you tried using the table definition that comes with the Asterisk
> source?
> 
> the file mysql_config.sql is located in contrib/realtime/mysql and defines
> a very different voicemail table than what you have in your configuration.
> 
> On Tue, May 3, 2016 at 3:10 AM, Michele Pinassi <michele.pina...@unisi.it>
> wrote:
> 
>> Hi all,
>> 
>> i'm experiencing a really frustrating issue with my Asterisk 13.7.2 with
>> realtime configuration on MySQL and Voicemail.
>> 
>> Here's res_config_mysql.conf:
>> 
>> *[default]*
>> *dbhost = 192.168.1.1*
>> *dbname = asterisk*
>> *dbuser = asterisk*
>> *dbpass = [x]*
>> *dbport = 3306*
>> *requirements=warn ; or createclose or createchar*
>> 
>> extconfig.conf:
>> 
>> *[settings]*
>> *sipusers => mysql,default,sipusers*
>> *sippeers => mysql,default,sipusers*
>> *sipregs => mysql,default,sipregs*
>> *voicemail => mysql,default,vmusers*
>> *meetme => mysql,default,meetme*
>> 
>> on Asterisk console:
>> 
>> *asterisk*CLI> realtime mysql status *
>> *default connected to asterisk@192.168.1.1 <asterisk@192.168.1.1>, port
>> 3306 with username asterisk for 56 minutes.*
>> *asterisk*CLI> *
>> 
>> "vmusers" table on MySQL:
>> 
>> 
>> uniqueid
>> <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk=vmusers_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60uniqueid%60+ASC_max_rows=25=81771f45cae5714ad1fac75365e0e494>
>> customer_id
>> <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk=vmusers_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60customer_id%60+ASC_max_rows=25=81771f45cae5714ad1fac75365e0e494>
>> context
>> <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk=vmusers_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60context%60+ASC_max_rows=25=81771f45cae5714ad1fac75365e0e494>
>> mailbox
>> <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk

Re: [asterisk-users] Double queue calls being delivered to agents

[Derek Bolichowski]

Looks like it missed 13.9.0 ☹

Thanks,
Derek B.

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Mudgett
Sent: Wednesday, May 04, 2016 10:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] Double queue calls being delivered to agents



On Tue, May 3, 2016 at 8:59 PM, Richard Mudgett 
<rmudg...@digium.com<mailto:rmudg...@digium.com>> wrote:


On Tue, May 3, 2016 at 6:15 PM, Derek Bolichowski 
<de...@empire-team.com<mailto:de...@empire-team.com>> wrote:
I posted this over in asterisk-dev, realized I probably should have put it here.

Hi there,
We’ve been having a strange issue with a customer’s queues where a queued call 
will ring an available agent, agent answers, then a second or two later the 
agent is offered a second call which they cannot answer, since they’re already 
speaking with a client.

This in turn causes a few issues:
- Agent stats are no longer accurate, as it gets marked down as a ‘missed call’.
- Cannot use ‘autopause’ feature any longer, as the second queue call goes 
unanswered and pauses the agents.

The basic queue setup is as follows:
Autofill = yes
Ringinuse = no
Wrapuptime = 5
Strategy = fewestcalls (tried ‘random’ also)
Timeout = 15

We’re on Asterisk 11.21.2 currently.

In talking to a few colleagues, they seem to recall there being an old patch 
for the Asterisk queues application that inserted a short 100ms delay between 
delivering first and second calls.  I’ve scoured the web today, and found some 
old forums posts of people looking for something exactly like this, but haven’t 
found the actual patch, if one even exists.

I’m hoping someone may have some suggestions on some options we can try to 
eliminate this issue.

Thanks for taking the time to read this.

This issue has been around a long time and was just recently fixed and I think
it was just released in the latest v11 version.
See https://issues.asterisk.org/jira/browse/ASTERISK-16115

Looks like it will be in the next release as the issue does not have a target 
release set.
Richard

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Re: [asterisk-users] Calls are dropped after 15 minutes

[Derek Bolichowski]

Set session-timers=refuse in sip.conf and do a sip reload.
We had this problem with a handful of devices and this ultimately stopped the 
issue.


Thanks,
Derek B.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jamie Stapleton
Sent: Tuesday, August 02, 2016 10:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Calls are dropped after 15 minutes

SIP re-invite (http://www.voip-info.org/wiki/view/SIP+method+invite+re-invite)
may be an issue as well.



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Re: [asterisk-users] Registration server with PJSIP

[Derek Bolichowski]

Hi Leandro,
I believe if you check /usr/local/src/astersisk-13.9.1/contrib/mysql you will 
find a .SQL file that would build the default tables for you.

Looking in the file, it appears there is a table created called `sippeers` 
which has a column `regserver VARCHAR(20),`

It will also create the other PJSIP-related tables such as ps_endpoints, 
ps_aors, etc.

I could be mistaken but perhaps `sippeers` is the table you’re looking for.

Thanks
Derek B.

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini
Sent: July 2, 2016 2:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: [asterisk-users] Registration server with PJSIP

Hello,
I am moving from realtime chan_sip to pjsip and one of the problem I am facing 
is the lack of "sipregs". With chan_sip, when an extension registers, the 
server where it has registered to is stored in sipregs.

Is there something similar in pjsip? How can I find on which server the pjsip 
extension has registered to?

Leandro



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[asterisk-users] queue show - extensions in call going from (in use) to (not in use)

[Derek Bolichowski]

Running Asterisk 11.23.0 realtime currently, I've noticed some odd queue 
behavior only starting today. We are using queues in conjunction with FOP2 to 
show agent status, etc.  As of today, we noticed that an agent will be on a 
call but show as available.

On an inbound queue call, when an agent answers the output of `queue show` will 
look like this:

3103 (Local/AG-000-NF-1174@fromotherpbx/n from Custom:3103-tenant) (ringinuse 
disabled) (realtime) (in call) (In use) has taken 1 calls (last was 69 secs ago)

Sometimes, and not always, the call will turn into this, causing FOP to show 
the agent as available again:

3103 (Local/AG-000-NF-1174@fromotherpbx/n from Custom:3103-tenant) (ringinuse 
disabled) (realtime) (in call) (Not in use) has taken 1 calls (last was 282 
secs ago)

Out of curiosity, is there something that would make the queue app think that 
the extension is no longer in use?  My thought process would be that if the 
agent is in a call, that it should always be 'in use', but perhaps I'm missing 
some pertinent information here.

Any tips would be appreciated.

Thanks,
Derek B.

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Re: [asterisk-users] Dropped call after 900s: 481 call/transaction does not exist and another anomaly during re-invite in timer - full anonymized trace attached

[Derek Bolichowski]

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael Maier
Sent: Wednesday, November 30, 2016 12:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: [asterisk-users] Dropped call after 900s: 481 call/transaction does 
not exist and another anomaly during re-invite in timer - full anonymized trace 
attached

Hello all!

I can see a strange problem during invite in dialog in the context of timer 
handling.

Given is the following incoming call from provider at 8.195.88.234 (2@2) to my 
asterisk at 28.19.57.152 (1@1):

After 900s suddenly *asterisk* starts the timer reinvite - I would have 
expected the reinvite started by the provider as usual.

The expected reinvite by the provider is started during authentication of the 
reinvite started by asterisk and is answered immediately by asterisk with sip 
481.

The answer of the provider after the resend of the reinvite came about 0.5s 
later and is sip 481, too.

=> The session obviously isn't known on both sides!

Asterisk therefore now drops the call (bye).


Does anybody has any idea about the reason why both members don't recognize the 
existing session any more? I hope the attached sip trace can shed some light on 
the problem.


Thanks,
Michael


HI Michael,
You can set this in sip.conf:
session-timers=refuse

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Re: [asterisk-users] Asterisk 1.8.32.3 : no video (h.264)

[Derek Bolichowski]

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, April 21, 2017 10:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Asterisk 1.8.32.3 : no video (h.264)

Hello


you mean while placing a video call ? What info am I looking for in the debug 
output ?




Kind regards.

J.




Why not try removing all codecs from the SIP Peer (deny all, 
allow only H264), unregister the peer, and try a video call again?  If it 
works, try adding G711 back but keep H264 at the top of the priority.
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Re: [asterisk-users] Hack attempt sequential config file read looking for valid files.

[Derek Bolichowski]

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Friday, April 21, 2017 12:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: [asterisk-users] Hack attempt sequential config file read looking for 
valid files.

I "justed" happened to look at /var/log/messages...

I saw:
Apr 21 12:18:40 in.tftpd[22719]: RRQ from 69.64.57.18 filename 0004f2034f6b.cfg
Apr 21 12:18:40 in.tftpd[22719]: Client 69.64.57.18 File not found 
0004f2034f6b.cfg
Apr 21 12:18:40 in.tftpd[22720]: RRQ from 69.64.57.18 filename 0004f2034f6c.cfg
Apr 21 12:18:40 in.tftpd[22720]: Client 69.64.57.18 File not found 
0004f2034f6c.cfg
Apr 21 12:18:40 in.tftpd[22721]: RRQ from 69.64.57.18 filename 0004f2034f6d.cfg
Apr 21 12:18:40 in.tftpd[22721]: Client 69.64.57.18 File not found 
0004f2034f6d.cfg
Apr 21 12:18:40 in.tftpd[22722]: RRQ from 69.64.57.18 filename 0004f2034f6e.cfg

so basically an sequential read of polycom MAC address config files.
Some is trying to read to determine if I have any polycom files just sequential 
read after read.
And if so - it would get any extension and password at that time.
Luckily I have none.

However - how does one block attempts like this ?

Thanks!

Jerry


Jerry,
Can you change to FTP Provisioning, or HTTPS etc? Atleast with FTP you can set 
a user/pass to your directory with mac.cfg to prevent open access.
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