[asterisk-users] Double queue calls being delivered to agents
Derek Bolichowski
derek at empire-team.com
Wed May 4 18:45:15 CDT 2016
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-request at lists.digium.com" <asterisk-users-request at lists.digium.com> wrote:
>
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> Today's Topics:
>
> 1. Re: Asterisk 13 Realtime Voicemail frustrating issue
> (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 and php5.6
> (Mamadou NGOM)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Tue, 3 May 2016 11:39:44 -0700
> From: John Kiniston <johnkiniston at gmail.com>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Subject: Re: [asterisk-users] Asterisk 13 Realtime Voicemail
> frustrating issue
> Message-ID:
> <CAFJQOGc8SYL_fSL8PMr+p6F6P1-nzK-_3rAYraKW4kzJEv8AKA at 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.pinassi at 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 = [xxxxx]*
>> *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 at 192.168.1.1 <asterisk at 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&table=vmusers&sql_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60uniqueid%60+ASC&session_max_rows=25&token=81771f45cae5714ad1fac75365e0e494>
>> customer_id
>> <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk&table=vmusers&sql_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60customer_id%60+ASC&session_max_rows=25&token=81771f45cae5714ad1fac75365e0e494>
>> context
>> <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk&table=vmusers&sql_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60context%60+ASC&session_max_rows=25&token=81771f45cae5714ad1fac75365e0e494>
>> mailbox
>> <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk&table=vmusers&sql_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60mailbox%60+ASC&session_max_rows=25&token=81771f45cae5714ad1fac75365e0e494>
>> password
>> <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk&table=vmusers&sql_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60password%60+ASC&session_max_rows=25&token=81771f45cae5714ad1fac75365e0e494>
>> fullname
>> <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk&table=vmusers&sql_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60fullname%60+ASC&session_max_rows=25&token=81771f45cae5714ad1fac75365e0e494>
>> email
>> <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk&table=vmusers&sql_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60email%60+ASC&session_max_rows=25&token=81771f45cae5714ad1fac75365e0e494>
>> pager
>> <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk&table=vmusers&sql_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60pager%60+ASC&session_max_rows=25&token=81771f45cae5714ad1fac75365e0e494>
>> stamp
>> <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk&table=vmusers&sql_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60stamp%60+DESC&session_max_rows=25&token=81771f45cae5714ad1fac75365e0e494>
>> 5002 5002 default 5002 xxxx AAA
>>
>> *NULL* 0000-00-00 00:00:00
>> 5005 5005 default 5005 xxxx bbb
>> *NULL* 0000-00-00 00:00:00
>> 5018 5018 default 5018 xxxx ccc
>> *NULL* 0000-00-00 00:00:00
>> 5007 5007 default 5007 xxxx sdddd
>> *NULL* 0000-00-00 00:00:00
>> *BUT* when i type, on Asterisk console:
>>
>> *asterisk*CLI> voicemail show zones *
>> *There are no voicemail zones currently defined*
>> *Command 'voicemail show zones ' failed.*
>> *asterisk*CLI> *
>>
>> the same, of course, for "show users default". And whet i try to access a
>> mailbox, i get a "Invalid password".
>>
>> Any hints ? Please, i'm really frustrated !
>>
>> Michele
>>
>> --
>> Michele Pinassi
>> Responsabile Telefonia di Ateneo
>> Servizio Reti, Sistemi e Sicurezza Informatica - Universit? degli Studi di Siena
>> tel: 0577.(23)5000 - centralino at unisi.it
>>
>> Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di Ateneo, http://www.faq.unisi.it
>>
>>
>> --
>> _____________________________________________________________________
>> -- 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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>
>
>
> --
> A human being should be able to change a diaper, plan an invasion, butcher
> a hog, conn a ship, design a building, write a sonnet, balance accounts,
> build a wall, set a bone, comfort the dying, take orders, give orders,
> cooperate, act alone, solve equations, analyze a new problem, pitch manure,
> program a computer, cook a tasty meal, fight efficiently, die gallantly.
> Specialization is for insects.
> ---Heinlein
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> ------------------------------
>
> Message: 2
> Date: Tue, 3 May 2016 20:45:05 +0200
> From: Michael Maier <m1278468 at allmail.net>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Subject: Re: [asterisk-users] Migrating asterisk 11 to 13: some
> callers get no ringback tone any more
> Message-ID: <5728F1B1.5030102 at allmail.net>
> Content-Type: text/plain; charset=windows-1252
>
>> On 05/03/2016 at 05:43 PM Joshua Colp wrote:
>> Michael Maier wrote:
>>>> On 05/03/2016 at 04:50 PM Joshua Colp wrote:
>>>> Michael Maier wrote:
>>>>> Hello Joshua!
>>>>>
>>>>>
>>>>> I attached the sip debug without the progressinband=never set. The
>>>>> caller didn't get a ring back tone as expected.
>>>> Please keep this on list so that anyone who may run into a similar
>>>> problem in the future has a chance of finding this discussion.
>>>
>>> You are right - normally I'm going exactly this way. But I don't want
>>> the traces to be world wide readable (-> privacy). I will write a
>>> summary to the list as far as we know more.
>>>
>>>> As for your log there's nothing of note really, it's just expecting to
>>>> send the ringing as inband audio instead of out of band. Does "rtp set
>>>> debug on" show the RTP traffic going to the other side?
>>>
>>> Yes. I attached it.
>>>
>>> And no - there isn't any packet blocked by iptables :-).
>>
>> There is nothing abnormal here and Asterisk appears to be doing the
>> correct thing. It's sending an audio stream with early progress to the
>> caller. It may be that in a previous FreePBX, or when used with 13, they
>> changed the behavior for this to force early media and the provider is
>> not allowing it.
>
> Ok - but this doesn't seem to answer my main question:
>
> Why must
>
> progressinband=never
>
> be applied especially if asterisk uses it by default? The big difference
> between w/ and w/o it is:
>
> w/o the option progrssinband=never just the sip-package
> 183 Session Progress
> is sent.
>
> w/ the option set, the additional sip-packages
> 100 Trying
> 180 Ringing
> 180 Ringing
> are sent.
>
> If progrssinband=never is the default, the Ringing package should be
> sent always, shouldn't it?
>
> If I remove the option progrssinband=never via FreePBX, I can't find any
> other value provided to progrssinband in /etc/asterisk/*.
>
>
> Why does it work always correctly w/ the second trunk, which is
> connected directly to the extension?
>
> Is it possible to switch off the standard behavior of asterisk /
> progrssinband for ring groups only by setting some other options?
>
>
>
> Thanks,
> kind regards,
> Michael
>
>
>
> ------------------------------
>
> Message: 3
> Date: Tue, 03 May 2016 15:52:05 -0300
> From: Joshua Colp <jcolp at digium.com>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Subject: Re: [asterisk-users] Migrating asterisk 11 to 13: some
> callers get no ringback tone any more
> Message-ID: <5728F355.1020703 at digium.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Whoops, email client auto-filled dev previously. Let's try this again.
>
> Michael Maier wrote:
>
> <snip>
>
>> Ok - but this doesn't seem to answer my main question:
>>
>> Why must
>>
>> progressinband=never
>>
>> be applied especially if asterisk uses it by default? The big difference
>> between w/ and w/o it is:
>
> The default in 13 is "no" which still allows early media through. That
> option has a complicated past.
>
>>
>> w/o the option progrssinband=never just the sip-package
>> 183 Session Progress
>> is sent.
>
> Yes, because it's doing inband progress using a media stream.
>
>>
>> w/ the option set, the additional sip-packages
>> 100 Trying
>> 180 Ringing
>> 180 Ringing
>> are sent.
>>
>> If progrssinband=never is the default, the Ringing package should be
>> sent always, shouldn't it?
>
> It's not the default.
>
>>
>> If I remove the option progrssinband=never via FreePBX, I can't find any
>> other value provided to progrssinband in /etc/asterisk/*.
>>
>>
>> Why does it work always correctly w/ the second trunk, which is
>> connected directly to the extension?
>
> FreePBX may not use inband progress for that scenario, causing it to
> send out of band ringing instead.
>
>>
>> Is it possible to switch off the standard behavior of asterisk /
>> progrssinband for ring groups only by setting some other options?
>
> Asterisk does not have the concept of ring groups, this is a FreePBX
> construct. Asterisk itself does allow the option to be set on an
> individual basis for the entries in sip.conf.
>
> --
> 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
>
>
>
>
> ------------------------------
>
> Message: 4
> Date: Tue, 3 May 2016 15:07:09 -0400
> From: Eric Wieling <ewieling at nyigc.com>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Subject: Re: [asterisk-users] Migrating asterisk 11 to 13: some
> callers get no ringback tone any more
> Message-ID: <5728F6DD.9000907 at nyigc.com>
> Content-Type: text/plain; charset=windows-1252; format=flowed
>
> I don't know the default setting for progressinband in the code, but it
> is documented in Asterisk 11's sip.conf.sample as defaulting to never.
> Maybe the docs were fixed since Asterisk 11.
>
> from 11.21.x sip.conf.sample:
>
> ;progressinband=never ; If we should generate in-band ringing
> always
> ; use 'never' to never use in-band
> signalling, even in cases
> ; where some buggy devices might not
> render it
> ; Valid values: yes, no, never Default:
> never
>
>
>> On 05/03/2016 02:52 PM, Joshua Colp wrote:
>> Whoops, email client auto-filled dev previously. Let's try this again.
>>
>> Michael Maier wrote:
>>
>> <snip>
>>
>>> Ok - but this doesn't seem to answer my main question:
>>>
>>> Why must
>>>
>>> progressinband=never
>>>
>>> be applied especially if asterisk uses it by default? The big
>> difference
>>> between w/ and w/o it is:
>>
>> The default in 13 is "no" which still allows early media through. That
>> option has a complicated past.
>>
>>>
>>> w/o the option progrssinband=never just the sip-package
>>> 183 Session Progress
>>> is sent.
>>
>> Yes, because it's doing inband progress using a media stream.
>>
>>>
>>> w/ the option set, the additional sip-packages
>>> 100 Trying
>>> 180 Ringing
>>> 180 Ringing
>>> are sent.
>>>
>>> If progrssinband=never is the default, the Ringing package should be
>>> sent always, shouldn't it?
>>
>> It's not the default.
>>
>>>
>>> If I remove the option progrssinband=never via FreePBX, I can't find
>> any
>>> other value provided to progrssinband in /etc/asterisk/*.
>>>
>>>
>>> Why does it work always correctly w/ the second trunk, which is
>>> connected directly to the extension?
>>
>> FreePBX may not use inband progress for that scenario, causing it to
>> send out of band ringing instead.
>>
>>>
>>> Is it possible to switch off the standard behavior of asterisk /
>>> progrssinband for ring groups only by setting some other options?
>>
>> Asterisk does not have the concept of ring groups, this is a FreePBX
>> construct. Asterisk itself does allow the option to be set on an
>> individual basis for the entries in sip.conf.
>
> --
> if at first you don't succeed, skydiving isn't for you
>
>
>
>
> ------------------------------
>
> Message: 5
> Date: Tue, 03 May 2016 16:16:04 -0300
> From: Joshua Colp <jcolp at digium.com>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Subject: Re: [asterisk-users] Migrating asterisk 11 to 13: some
> callers get no ringback tone any more
> Message-ID: <5728F8F4.4080502 at digium.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Eric Wieling wrote:
>> I don't know the default setting for progressinband in the code, but it
>> is documented in Asterisk 11's sip.conf.sample as defaulting to never.
>> Maybe the docs were fixed since Asterisk 11.
>
> The behavior change to actually do what the option was documented to do.
> As part of that the default was changed to reflect the past behavior,
> thus why it was changed to no. The commit itself:
>
> chan_sip: make progressinband default to no
>
> After the "progressinband" value setting of "never" was updated to never
> send a 183 this separated its use from the "no" value. Since "never" was
> the default, but most users probably expect "no" this patch updates the
> default for the "progressinband" setting to "no."
>
> This was tracked under ASTERISK-24835[1].
>
> [1] https://issues.asterisk.org/jira/browse/ASTERISK-24835
>
> --
> 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
>
>
>
>
> ------------------------------
>
> Message: 6
> Date: Tue, 3 May 2016 14:24:25 -0700
> From: John Kiniston <johnkiniston at gmail.com>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Subject: [asterisk-users] Call a subroutine via Originate?
> Message-ID:
> <CAFJQOGf1kx+yfP+0xk1j8kLmqWTNXQ5r_zkrVusPb4VFUQ67vg at mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> Howdy everyone,
>
> I'm writing a little click to dial type tool and I've run into a snag where
> my Originate command needs to call a Sub routine to do a database lookup
> and some other stuff.
>
> I can't seem to get the syntax right to call Gosub with Originate
>
> Just testing with the command line I've been unable to make it work with
> any of these attempts:
>
> originate PJSIP/johntest application Gosub sub-callout s,1
>
> originate PJSIP/johntest application Gosub sub-callout(s,1)
>
> originate PJSIP/johntest application Gosub (sub-callout,s,1)
>
> What Syntax should I be using?
>
> And if it helps I'll be calling this via AMI over https.
>
> Thanks!
>
> --
> A human being should be able to change a diaper, plan an invasion, butcher
> a hog, conn a ship, design a building, write a sonnet, balance accounts,
> build a wall, set a bone, comfort the dying, take orders, give orders,
> cooperate, act alone, solve equations, analyze a new problem, pitch manure,
> program a computer, cook a tasty meal, fight efficiently, die gallantly.
> Specialization is for insects.
> ---Heinlein
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> ------------------------------
>
> Message: 7
> Date: Tue, 3 May 2016 14:33:32 -0700
> From: Bruce Ferrell <bferrell at baywinds.org>
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] Call a subroutine via Originate?
> Message-ID: <9bcf1278-a6c1-b3e5-668d-fd4cdb3f371c at baywinds.org>
> Content-Type: text/plain; charset="windows-1252"; Format="flowed"
>
> use the macro construct and return from the macro
>
>> On 5/3/16 2:24 PM, John Kiniston wrote:
>> Howdy everyone,
>>
>> I'm writing a little click to dial type tool and I've run into a snag
>> where my Originate command needs to call a Sub routine to do a
>> database lookup and some other stuff.
>>
>> I can't seem to get the syntax right to call Gosub with Originate
>>
>> Just testing with the command line I've been unable to make it work
>> with any of these attempts:
>>
>> originate PJSIP/johntest application Gosub sub-callout s,1
>>
>> originate PJSIP/johntest application Gosub sub-callout(s,1)
>>
>> originate PJSIP/johntest application Gosub (sub-callout,s,1)
>>
>> What Syntax should I be using?
>>
>> And if it helps I'll be calling this via AMI over https.
>>
>> Thanks!
>>
>> --
>> A human being should be able to change a diaper, plan an invasion,
>> butcher a hog, conn a ship, design a building, write a sonnet, balance
>> accounts, build a wall, set a bone, comfort the dying, take orders,
>> give orders, cooperate, act alone, solve equations, analyze a new
>> problem, pitch manure, program a computer, cook a tasty meal, fight
>> efficiently, die gallantly. Specialization is for insects.
>> ---Heinlein
>
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> ------------------------------
>
> Message: 8
> Date: Tue, 3 May 2016 23:15:53 +0000
> From: Derek Bolichowski <derek at empire-team.com>
> To: "asterisk-users at lists.digium.com"
> <asterisk-users at lists.digium.com>
> Subject: [asterisk-users] Double queue calls being delivered to agents
> Message-ID: <DA688326-8EDC-470F-9355-F4E5612B0868 at empire-team.com>
> Content-Type: text/plain; charset="utf-8"
>
> 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
>
> ------------------------------
>
> Message: 9
> Date: Tue, 3 May 2016 19:48:25 -0400
> From: Saint Michael <venefax at gmail.com>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Subject: [asterisk-users] Execute an app on the master channel from
> inside a Macro on the called channel
> Message-ID:
> <CAC9cSOBWVp0gyibM+skTWnt3yG6oThV8WBwqLgLgjAwm2UXCfA at mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> ?While I am executing a Macro on the called channel, right after the call
> connects?, I need to execute an app on the master channel, from inside that
> macro, specifically, SendDTMF. If I execute it now, it send a text message
> to the Callee, when my app needs to send it to the caller.
>
> I could use
> set(master_channel(variable)=XXX), but then how do I execute some code on
> the master channel.
> Note that I could send the name of the master channels to the Macro
> M(Name^parameter), but then how do I execute SendDtmf on the identified
> Master Channel?
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> ------------------------------
>
> Message: 10
> Date: Tue, 3 May 2016 20:59:14 -0500
> From: Richard Mudgett <rmudgett at digium.com>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Subject: Re: [asterisk-users] Double queue calls being delivered to
> agents
> Message-ID:
> <CALD46g0MUWkRyXCokM7Kf2nZO+rN3GzAwcScnk_sirJBpR2dQg at mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> On Tue, May 3, 2016 at 6:15 PM, Derek Bolichowski <derek at 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
>
> Richard
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> ------------------------------
>
> Message: 11
> Date: Wed, 4 May 2016 09:09:17 +0200
> From: Michael Maier <m1278468 at allmail.net>
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] Migrating asterisk 11 to 13: some
> callers get no ringback tone any more
> Message-ID: <5729A01D.7030502 at allmail.net>
> Content-Type: text/plain; charset=windows-1252
>
>> On 05/03/2016 at 09:16 PM Joshua Colp wrote:
>> Eric Wieling wrote:
>>> I don't know the default setting for progressinband in the code, but it
>>> is documented in Asterisk 11's sip.conf.sample as defaulting to never.
>>> Maybe the docs were fixed since Asterisk 11.
>>
>> The behavior change to actually do what the option was documented to do.
>> As part of that the default was changed to reflect the past behavior,
>> thus why it was changed to no. The commit itself:
>>
>> chan_sip: make progressinband default to no
>>
>> After the "progressinband" value setting of "never" was updated to never
>> send a 183 this separated its use from the "no" value.
>
> But "never" option therefore sends 180 Ringing which I was missing. The
> new default "no" doesn't send 180 Ringing any more ... .
>
>> Since "never" was
>> the default, but most users probably expect "no" this patch updates the
>> default for the "progressinband" setting to "no."
>>
>> This was tracked under ASTERISK-24835[1].
>>
>> [1] https://issues.asterisk.org/jira/browse/ASTERISK-24835
>
> This makes sense! I migrated from
>
> asterisk11-11.8.1-40_centos6.x86_64,
>
> which had the default progressinband=never to
>
> asterisk13-core-13.7.2-1.shmz65.1.94.x86_64
>
> which had the new default.
>
> POTS callers advertise support for early media - mobile callers on the
> other hand don't advertise it, therefore mobile wasn't a problem because
> early media (183) isn't triggered (and used!) at all.
>
>
> Two strange things being left:
>
> 1. Why does progressinband=no work, if there is *no* ringgroup between
> trunk and extension. This seems to be a "feature" of FreePBX.
>
> 2. Why is early media used even if the caller doesn't advertise it? Are
> there other triggers like P-Early-Media?
>
>
>
>
> Another basic question:
> What do I need early media exactly for? I'm only using SIP phones -
> nothing else. Couldn't it be completely disabled for these trunks? Or
> would it break things like voice mail service e.g.? How can I disable it
> completely even if it is advertised by the caller?
>
>
> Thanks,
> Michael
>
>
>
> ------------------------------
>
> Message: 12
> Date: Wed, 4 May 2016 11:12:27 +0200
> From: Olivier <oza.4h07 at gmail.com>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Subject: Re: [asterisk-users] T.38 with Audiocodes gateway [SOLVED]
> Message-ID:
> <CAPeT9jgOwcuOa-jyErR9b1+FWF_-kfxUQFX74EMAkzUQUy+nug at mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> 2016-05-03 16:43 GMT+02:00 Matt Fredrickson <creslin at digium.com>:
>
>>> On Fri, Apr 29, 2016 at 1:34 AM, Olivier <oza.4h07 at gmail.com> wrote:
>>> Hello,
>>>
>>> I'm helping a colleague (*) which has the following setup:
>>>
>>> ITSP --- <T.38 capable PJSIP trunk> --- Asterisk 13 --- <PJSIP>--
>>> Audiocodes MP-112 --- <FXO/FXS> --- Fax machine
>>>
>>> My issue is the following :
>>> Audiocodes gateway reject INVITEs with 488 Not Acceptable Here
>>>
>>> It seems this gateway requires t38 settings to be present in SDP body in
>> the
>>> very first INVITE.
>>>
>>> My questions are the following:
>>>
>>> 1. I expected T.38 to exclusively work with reINVITE where calls are
>>> established as normal voice calls (PCMA/PCMU in SDP, for instance) and
>> then
>>> upgraded to T.38 (when CNG is detected, for instance).
>>> Have you ever heard of T.38 sessions being established right from the
>> start
>>> (ie with T.38 settings in the first INVITE) ?
>>
>> No. It would seem to be extremely broken if it denies a call based on
>> a lack of T.38 sdp parameters on the initial INVITE.
>
> OK
>
>>
>>> 2. Is it possible to configure Asterisk to pass T.38 settings in SDP in
>> the
>>> first INVITE it sends ?
>>>
>>> 3. Any suggestion with Audiocodes gateway ?
>>
>> Look for T.38 settings maybe? See if there is something keeping you
>> from sending an initial invite with non-T.38 SDP....?
>
> Yes, I think issue must come from incorrect Audiocodes settings.
> Requiring T.38 settings within first INVITE seems very unusual.
>
> Thank you very much for replying
>
>>
>> --
>> Matthew Fredrickson
>> Digium, Inc. | Engineering Manager
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
>>
>> --
>> _____________________________________________________________________
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> ------------------------------
>
> Message: 13
> Date: Wed, 4 May 2016 13:25:53 +0200
> From: Sebastian Damm <damm at sipgate.de>
> To: Asterisk <asterisk-users at lists.digium.com>
> Subject: [asterisk-users] Asterisk registers with TLS, but sends out
> calls via UDP
> Message-ID:
> <CABkWSFwAk2jmf08zj-FSgMbgTV48rhKsQ0LVb-QrudOu1goRTA at mail.gmail.com>
> Content-Type: text/plain; charset=UTF-8
>
> Hi,
>
> I have an Asterisk 13.8.2, which is supposed to be only a client to an
> encrypted SIP service. All local phones are connected via UDP.
>
> Since I can't use PJSIP (see my mailing list post from yesterday), I
> tried configuring chan_sip to work that way. My settings are:
>
> [general]
> context=public
> allowoverlap=no
> udpbindaddr=0.0.0.
> tlsbindaddr=0.0.0.0
> tcpenable=yes
> tcpbindaddr=0.0.0.0
> tlsenable=yes
> transport=udp
> srvlookup=yes
> tlscafile=/usr/local/etc/asterisk/keys/4cfd3c78.0
> tlscapath=/usr/local/etc/asterisk/keys
> tlsclientmethod=tlsv1
> sipdebug = yes
>
> register => tls://1234567@example.org:foobar@dev.example.org
>
> [devtrunk]
> type=peer
> host=example.org
> defaultuser=1234567
> fromuser=1234567
> remotesecret=foobar
> transport=tls
> outboundproxy=dev.example.org
> context=carrier-in
> encryption=yes
>
> When I start up, I see my Asterisk doing a _sips._tcp SRV lookup, but
> that's just for the registration, I guess. I also see it doing
> _sip._udp SRV queries. I wouldn't know why it would have to do that.
>
> The REGISTER packets are sent out via TLS, as I would expect.
>
> When I issue a "sip show peer devtrunk" command, it tells me this:
>
> Prim.Transp. : TLS
> Allowed.Trsp : TLS
>
> Looks okay to me. But when I place a call, Asterisk does this:
>
> Reliably Transmitting (no NAT) to 2.3.4.5:5060:
> INVITE sip:0123456789 at example.org SIP/2.0
> Via: SIP/2.0/UDP 9.8.7.6:0;branch=z9hG4bK2974d534
>
> It sends the packet out via UDP, and to the wrong host, since it
> doesn't use the correct SRV entry and instead sends it to the UDP
> server.
>
> I did not generate a certificate for my Asterisk, because it only acts
> as a client. I think, this shouldn't be needed.
>
> Can anyone point me to where I misconfigured something? Or did I
> stumble upon a bug? What would I have to do to make Asterisk use the
> open TLS connection used for registering for outbound calls, too?
>
> Best Regards,
> Sebastian
>
>
>
> ------------------------------
>
> Message: 14
> Date: Wed, 4 May 2016 14:49:34 +0200 (CEST)
> From: Mamadou NGOM <ngom at numericap.com>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Subject: [asterisk-users] Compatibilty between agi for asterisk 13.8.0
> and php5.6
> Message-ID:
> <1979110061.191209.7cc1d90d-d410-4a02-a619-42e64003d44e.open-xchange at email.1and1.fr>
>
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