derek at empire-team.com
Guest
|
 Posted: Wed May 04, 2016 6:45 pm Post subject: [asterisk-users] Double queue calls being delivered to agen |
 |
|
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
| Quote: | On May 4, 2016, at 8:50 AM, "asterisk-users-request@lists.digium.com" <asterisk-users-request@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@gmail.com>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] Asterisk 13 Realtime Voicemail
frustrating issue
Message-ID:
<CAFJQOGc8SYL_fSL8PMr+p6F6P1-nzK-_3rAYraKW4kzJEv8AKA@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@unisi.it>
wrote:
| Quote: | 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@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&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@unisi.it
Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di Ateneo, http://www.faq.unisi.it
--
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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@allmail.net>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] Migrating asterisk 11 to 13: some
callers get no ringback tone any more
Message-ID: <5728F1B1.5030102@allmail.net>
Content-Type: text/plain; charset=windows-1252
| Quote: | On 05/03/2016 at 05:43 PM Joshua Colp wrote:
Michael Maier wrote:
| Quote: | | Quote: | On 05/03/2016 at 04:50 PM Joshua Colp wrote:
Michael Maier wrote:
| Quote: | 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.
| Quote: | 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@digium.com>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] Migrating asterisk 11 to 13: some
callers get no ringback tone any more
Message-ID: <5728F355.1020703@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>
| Quote: | 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.
| Quote: |
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.
| Quote: |
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.
| Quote: |
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.
| Quote: |
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@nyigc.com>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] Migrating asterisk 11 to 13: some
callers get no ringback tone any more
Message-ID: <5728F6DD.9000907@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
| Quote: | 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>
| Quote: | 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
| Quote: | 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.
| Quote: |
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.
| Quote: |
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.
| Quote: |
If I remove the option progrssinband=never via FreePBX, I can't find
|
any
| Quote: | 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.
| Quote: |
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@digium.com>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] Migrating asterisk 11 to 13: some
callers get no ringback tone any more
Message-ID: <5728F8F4.4080502@digium.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Eric Wieling wrote:
| Quote: | 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@gmail.com>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Subject: [asterisk-users] Call a subroutine via Originate?
Message-ID:
<CAFJQOGf1kx+yfP+0xk1j8kLmqWTNXQ5r_zkrVusPb4VFUQ67vg@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@baywinds.org>
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Call a subroutine via Originate?
Message-ID: <9bcf1278-a6c1-b3e5-668d-fd4cdb3f371c@baywinds.org>
Content-Type: text/plain; charset="windows-1252"; Format="flowed"
use the macro construct and return from the macro
| Quote: | 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@empire-team.com>
To: "asterisk-users@lists.digium.com"
<asterisk-users@lists.digium.com>
Subject: [asterisk-users] Double queue calls being delivered to agents
Message-ID: <DA688326-8EDC-470F-9355-F4E5612B0868@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@gmail.com>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@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@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@digium.com>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] Double queue calls being delivered to
agents
Message-ID:
<CALD46g0MUWkRyXCokM7Kf2nZO+rN3GzAwcScnk_sirJBpR2dQg@mail.gmail.com>
Content-Type: text/plain; charset="utf-8"
On Tue, May 3, 2016 at 6:15 PM, Derek Bolichowski <derek@empire-team.com>
wrote:
| Quote: | 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@allmail.net>
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Migrating asterisk 11 to 13: some
callers get no ringback tone any more
Message-ID: <5729A01D.7030502@allmail.net>
Content-Type: text/plain; charset=windows-1252
| Quote: | On 05/03/2016 at 09:16 PM Joshua Colp wrote:
Eric Wieling wrote:
| Quote: | 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 ... .
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@gmail.com>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] T.38 with Audiocodes gateway [SOLVED]
Message-ID:
<CAPeT9jgOwcuOa-jyErR9b1+FWF_-kfxUQFX74EMAkzUQUy+nug@mail.gmail.com>
Content-Type: text/plain; charset="utf-8"
2016-05-03 16:43 GMT+02:00 Matt Fredrickson <creslin@digium.com>:
| Quote: | | Quote: | On Fri, Apr 29, 2016 at 1:34 AM, Olivier <oza.4h07@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
| Quote: | 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
| Quote: | upgraded to T.38 (when CNG is detected, for instance).
Have you ever heard of T.38 sessions being established right from the
|
start
| Quote: | (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
| Quote: |
| Quote: | 2. Is it possible to configure Asterisk to pass T.38 settings in SDP in
|
the
| Quote: | 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
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Message: 13
Date: Wed, 4 May 2016 13:25:53 +0200
From: Sebastian Damm <damm@sipgate.de>
To: Asterisk <asterisk-users@lists.digium.com>
Subject: [asterisk-users] Asterisk registers with TLS, but sends out
calls via UDP
Message-ID:
<CABkWSFwAk2jmf08zj-FSgMbgTV48rhKsQ0LVb-QrudOu1goRTA@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@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@numericap.com>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@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@email.1and1.fr>
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