Re: [asterisk-users] Realtime PJSIP RTT

[Joshua C. Colp]

On Tue, May 26, 2020 at 12:02 PM Nick Olsen <
n...@floridavirtualsolutions.com> wrote:

> Thanks Joshua, I assume by query asterisk you mean I'll need to query it
> via AMI? Is that information available via AMI
> ?
>

There is the PJSIPShowContacts AMI action[1] which returns ContactList
events[2] that contain the RTT. The ContactStatus event[3] is also raised
when things change if the connection is persistent.

[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+ManagerAction_PJSIPShowContacts
[2]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+ManagerEvent_ContactList
[3]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+ManagerEvent_ContactStatus

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
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Re: [asterisk-users] Realtime PJSIP RTT

[Nick Olsen]

Thanks Joshua, I assume by query asterisk you mean I'll need to query it
via AMI? Is that information available via AMI?

*Nick Olsen*
Network Engineer
Office: 321-408-5000 x103
Mobile: 321-794-0763



On Tue, May 26, 2020 at 2:57 PM Joshua C. Colp  wrote:

> On Tue, May 26, 2020 at 10:48 AM Nick Olsen <
> n...@floridavirtualsolutions.com> wrote:
>
>> Hello all,
>>
>> We would like to pull the RTT of registered endpoints from MySQL for use
>> in a webportal. However it doesn't appear asterisk tables this by default
>> like chan_sip did.
>>
>> I've found some information [1] that a modification of sorcery.conf can
>> get it writing to a table. But I'm struggling to figure out what that
>> configuration is. It seems the object is called "contact_status", and I've
>> tried [2] in sorcery.conf, But I'm not seeing any output or seeing anything
>> else to follow with GoogleFu.
>>
>> [1], A mailinglist crawler, i know:
>> https://asteriskfaqs.org/2015/10/29/asterisk-users/pjsip-and-rtt-in-realtime.html
>> [2], sourcery.conf
>> [res_pjsip]
>> **stuff***
>> contact_status=realtime,ps_contact_status (Which I've also defined in
>> extconfig
>> )
>>
>
> Contact status is not a sorcery object and is kept strictly in memory.
> There's no ability to store the information in a database, you'd need to
> query Asterisk.
>
> --
> Joshua C. Colp
> Asterisk Technical Lead
> Sangoma Technologies
> Check us out at www.sangoma.com and www.asterisk.org
> --
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> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Realtime PJSIP RTT

[Joshua C. Colp]

On Tue, May 26, 2020 at 10:48 AM Nick Olsen <
n...@floridavirtualsolutions.com> wrote:

> Hello all,
>
> We would like to pull the RTT of registered endpoints from MySQL for use
> in a webportal. However it doesn't appear asterisk tables this by default
> like chan_sip did.
>
> I've found some information [1] that a modification of sorcery.conf can
> get it writing to a table. But I'm struggling to figure out what that
> configuration is. It seems the object is called "contact_status", and I've
> tried [2] in sorcery.conf, But I'm not seeing any output or seeing anything
> else to follow with GoogleFu.
>
> [1], A mailinglist crawler, i know:
> https://asteriskfaqs.org/2015/10/29/asterisk-users/pjsip-and-rtt-in-realtime.html
> [2], sourcery.conf
> [res_pjsip]
> **stuff***
> contact_status=realtime,ps_contact_status (Which I've also defined in
> extconfig
> )
>

Contact status is not a sorcery object and is kept strictly in memory.
There's no ability to store the information in a database, you'd need to
query Asterisk.

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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[asterisk-users] Realtime PJSIP RTT

[Nick Olsen]

Hello all,

We would like to pull the RTT of registered endpoints from MySQL for use in
a webportal. However it doesn't appear asterisk tables this by default like
chan_sip did.

I've found some information [1] that a modification of sorcery.conf can get
it writing to a table. But I'm struggling to figure out what that
configuration is. It seems the object is called "contact_status", and I've
tried [2] in sorcery.conf, But I'm not seeing any output or seeing anything
else to follow with GoogleFu.

[1], A mailinglist crawler, i know:
https://asteriskfaqs.org/2015/10/29/asterisk-users/pjsip-and-rtt-in-realtime.html
[2], sourcery.conf
[res_pjsip]
**stuff***
contact_status=realtime,ps_contact_status (Which I've also defined in
extconfig)

*Nick Olsen*
Network Engineer
Office: 321-408-5000 x103
Mobile: 321-794-0763
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Re: [asterisk-users] Realtime PJSIP max_streams' issues

[Joshua C. Colp]

On Thu, Oct 24, 2019, at 3:28 PM, Ahmed Chohan wrote:
> 
> After altering the table; changing type from int(11) to varchar for 
> max_audio_streams & max_video_stream, it is working.

Can you please file an issue[1] for this with full details so we can look into 
what is precisely going on?

Thanks,

[1] https://issues.asterisk.org/jira

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

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Re: [asterisk-users] Realtime PJSIP max_streams' issues

[Ahmed Chohan]

After altering the table; changing type from int(11) to varchar for
max_audio_streams & max_video_stream, it is working.

Thanks Josh.

Date: Wed, 23 Oct 2019 14:40:48 -0300
> From: "Joshua C. Colp" 
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Realtime PJSIP max_streams' issues
> Message-ID: <28f0701f-3790-459b-8818-04c0975c6...@www.fastmail.com>
> Content-Type: text/plain
>
> On Wed, Oct 23, 2019, at 2:30 PM, Ahmed Chohan wrote:
> > The database I'm using is MySQL v 5.6.46.2, data type I'm using for
> > both parameters is int(11) the one created by the asterisk script; see
> > table structure below.
>
> If you alter it to be a varchar instead does that change the result within
> PJSIP?
>
> --
> Joshua C. Colp
> Digium - A Sangoma Company | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
>
> --
Regards,

Ahmed Munir Chohan
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Re: [asterisk-users] Realtime PJSIP max_streams' issues

[Joshua C. Colp]

On Wed, Oct 23, 2019, at 2:30 PM, Ahmed Chohan wrote:
> The database I'm using is MySQL v 5.6.46.2, data type I'm using for 
> both parameters is int(11) the one created by the asterisk script; see 
> table structure below.

If you alter it to be a varchar instead does that change the result within 
PJSIP?

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

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Re: [asterisk-users] Realtime PJSIP max_streams' issues

[Ahmed Chohan]

,
  `mwi_subscribe_replaces_unsolicited`
enum('0','1','off','on','false','true','no','yes') DEFAULT NULL,
  `deny` varchar(95) DEFAULT NULL,
  `permit` varchar(95) DEFAULT NULL,
  `acl` varchar(40) DEFAULT NULL,
  `contact_deny` varchar(95) DEFAULT NULL,
  `contact_permit` varchar(95) DEFAULT NULL,
  `contact_acl` varchar(40) DEFAULT NULL,
  `subscribe_context` varchar(40) DEFAULT NULL,
  `fax_detect_timeout` int(11) DEFAULT NULL,
  `contact_user` varchar(80) DEFAULT NULL,
  `preferred_codec_only` enum('yes','no') DEFAULT NULL,
  `asymmetric_rtp_codec` enum('yes','no') DEFAULT NULL,
  `rtcp_mux` enum('yes','no') DEFAULT NULL,
  `allow_overlap` enum('yes','no') DEFAULT NULL,
  `refer_blind_progress` enum('yes','no') DEFAULT NULL,
  `notify_early_inuse_ringing` enum('yes','no') DEFAULT NULL,
  `max_audio_streams` int(11) DEFAULT NULL,
  `max_video_streams` int(11) DEFAULT NULL,
  `webrtc` enum('yes','no') DEFAULT NULL,
  `dtls_fingerprint` enum('SHA-1','SHA-256') DEFAULT NULL,
  `incoming_mwi_mailbox` varchar(40) DEFAULT NULL,
  `bundle` enum('yes','no') DEFAULT NULL,
  `dtls_auto_generate_cert` enum('yes','no') DEFAULT NULL,
  `follow_early_media_fork` enum('yes','no') DEFAULT NULL,
  `accept_multiple_sdp_answers` enum('yes','no') DEFAULT NULL,
  `suppress_q850_reason_headers` enum('yes','no') DEFAULT NULL,
  `trust_connected_line` enum('0','1','off','on','false','true','no','yes')
DEFAULT NULL,
  `send_connected_line` enum('0','1','off','on','false','true','no','yes')
DEFAULT NULL,
  `ignore_183_without_sdp`
enum('0','1','off','on','false','true','no','yes') DEFAULT NULL,
  UNIQUE KEY `id` (`id`),
  KEY `ps_endpoints_id` (`id`)
) ENGINE=InnoDB DEFAULT CHARSET=latin1


-Original Message-
> From: asterisk-users  On Behalf
> Of Joshua C. Colp
> Sent: Tuesday, October 22, 2019 4:30 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Realtime PJSIP max_streams' issues
>
> On Tue, Oct 22, 2019, at 4:21 PM, Ahmed Chohan wrote:
> > Hi,
> >
> > I'm currently using Asterisk 16.4.0 cert version and working on webrtc.
> > For configuration perspective, I'm pretty much done with it but here
> > the real issue I'm currently facing i.e. when setting parameters
> > max_audio_streams & max_video_streams to any positive greater than 0
> > integer value in realtime (DB) of any endpoints. After running command
> > "pjsip show endpoint 100101" it shows '0' but when setting as
> > 'NULL' in DB, showing output to 1 for both parameters.
> >
> > Furthermore, in AOR section, the max_connection is set to 1 for each
> endpoints.
>
> The configuration option for there is max_contacts.
>
> > Please advise, for this issue.
>
> What database are you using? What type is the column? Do any other fields
> exhibit the problem?
>
> --
> Joshua C. Colp
> Digium - A Sangoma Company | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at:
> www.digium.com & www.asterisk.org
>
>
-- 
Regards,

Ahmed Munir Chohan
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Re: [asterisk-users] Realtime PJSIP max_streams' issues

[Dan Cropp]

Thanks Joshua.

This turned out to be my mistake.
Quiet variable was enabled on the User and needed to be disabled.

It's been at least a couple years since I wrote e-mails for my coworkers and 
forgot that setting.

Have a great day!
Dan

-Original Message-
From: asterisk-users  On Behalf Of 
Joshua C. Colp
Sent: Tuesday, October 22, 2019 4:30 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Realtime PJSIP max_streams' issues

On Tue, Oct 22, 2019, at 4:21 PM, Ahmed Chohan wrote:
> Hi,
> 
> I'm currently using Asterisk 16.4.0 cert version and working on webrtc. 
> For configuration perspective, I'm pretty much done with it but here 
> the real issue I'm currently facing i.e. when setting parameters 
> max_audio_streams & max_video_streams to any positive greater than 0 
> integer value in realtime (DB) of any endpoints. After running command 
> "pjsip show endpoint 100101" it shows '0' but when setting as 
> 'NULL' in DB, showing output to 1 for both parameters.
> 
> Furthermore, in AOR section, the max_connection is set to 1 for each 
> endpoints.

The configuration option for there is max_contacts.
 
> Please advise, for this issue.

What database are you using? What type is the column? Do any other fields 
exhibit the problem?

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

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_
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Re: [asterisk-users] Realtime PJSIP max_streams' issues

[Joshua C. Colp]

On Tue, Oct 22, 2019, at 4:21 PM, Ahmed Chohan wrote:
> Hi,
> 
> I'm currently using Asterisk 16.4.0 cert version and working on webrtc. 
> For configuration perspective, I'm pretty much done with it but here 
> the real issue I'm currently facing i.e. when setting parameters 
> max_audio_streams & max_video_streams to any positive greater than 0 
> integer value in realtime (DB) of any endpoints. After running command 
> "pjsip show endpoint 100101" it shows '0' but when setting as 
> 'NULL' in DB, showing output to 1 for both parameters.
> 
> Furthermore, in AOR section, the max_connection is set to 1 for each 
> endpoints.

The configuration option for there is max_contacts.
 
> Please advise, for this issue.

What database are you using? What type is the column? Do any other fields 
exhibit the problem?

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

-- 
_
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[asterisk-users] Realtime PJSIP max_streams' issues

[Ahmed Chohan]

Hi,

I'm currently using Asterisk 16.4.0 cert version and working on webrtc. For
configuration perspective, I'm pretty much done with it but here the real
issue I'm currently facing i.e. when setting parameters max_audio_streams &
max_video_streams to any positive greater than 0 integer value in realtime
(DB) of any endpoints. After running command "pjsip show endpoint
100101" it shows '0' but when setting as 'NULL' in DB, showing output
to 1 for both parameters.

Furthermore, in AOR section, the max_connection is set to 1 for each
endpoints.

Please advise, for this issue.

-- 
Regards,

Ahmed Munir Chohan
-- 
_
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Re: [asterisk-users] Realtime extensions, Multiple SQL lookups?

[Dovid Bender]

It's been years since I have used real time but can it be because of the
GoSub?



On Mon, Oct 16, 2017 at 12:39 PM, John Kiniston 
wrote:

> I'm toying with the idea of replacing a statically generated file I
> include in my extensions.conf with a realtime lookup against my database.
>
> I've got it working but something seems off in my logs, It looks like I'm
> getting two lookups for every priority?
>
> [Oct 12 16:45:24] DEBUG[26541][C-000e] res_odbc.c: Reusing ODBC handle
> 0x7f8da8002938 from class 'odbc_kiniston-test'
> [Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Skip: 0;
> SQL: SELECT * FROM extensions-test WHERE exten = ? AND priority = ? AND
> context = ?
> [Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Parameter 1
> ('exten') = '5206700792 <(520)%20670-0792>'
> [Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Parameter 2
> ('priority') = '5'
> [Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Parameter 3
> ('context') = 'sip-in'
> [Oct 12 16:45:24] DEBUG[26541][C-000e] res_odbc.c: Releasing ODBC
> handle 0x7f8da8002938 into pool
> [Oct 12 16:45:24] DEBUG[26541][C-000e] res_odbc.c: Reusing ODBC handle
> 0x7f8da8002938 from class 'odbc_kiniston-test'
> [Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Skip: 0;
> SQL: SELECT * FROM extensions-test WHERE exten = ? AND priority = ? AND
> context = ?
> [Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Parameter 1
> ('exten') = '5206700792 <(520)%20670-0792>'
> [Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Parameter 2
> ('priority') = '5'
> [Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Parameter 3
> ('context') = 'sip-in'
> [Oct 12 16:45:24] DEBUG[26541][C-000e] res_odbc.c: Releasing ODBC
> handle 0x7f8da8002938 into pool
> [Oct 12 16:45:24] VERBOSE[26541][C-000e] pbx_realtime.c: Executing
> [5206700792 <(520)%20670-0792>@sip-in:5] Gosub("PJSIP/trunks1-000e",
> "sub-setupinboundcall,s,1(kiniston,Pop_Country_Crossover,300,VOIP)")
>
> ---SNIP---
>
> [Oct 12 16:45:24] DEBUG[26541][C-000e] res_odbc.c: Reusing ODBC handle
> 0x7f8da8002938 from class 'odbc_kiniston-test'
> [Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Skip: 0;
> SQL: SELECT * FROM extensions-test WHERE exten = ? AND priority = ? AND
> context = ?
> [Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Parameter 1
> ('exten') = '5206700792 <(520)%20670-0792>'
> [Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Parameter 2
> ('priority') = '6'
> [Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Parameter 3
> ('context') = 'sip-in'
> [Oct 12 16:45:24] DEBUG[26541][C-000e] res_odbc.c: Releasing ODBC
> handle 0x7f8da8002938 into pool
> [Oct 12 16:45:24] DEBUG[26541][C-000e] res_odbc.c: Reusing ODBC handle
> 0x7f8da8002938 from class 'odbc_kiniston-test'
> [Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Skip: 0;
> SQL: SELECT * FROM extensions-test WHERE exten = ? AND priority = ? AND
> context = ?
> [Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Parameter 1
> ('exten') = '5206700792 <(520)%20670-0792>'
> [Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Parameter 2
> ('priority') = '6'
> [Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Parameter 3
> ('context') = 'sip-in'
> [Oct 12 16:45:24] DEBUG[26541][C-000e] res_odbc.c: Releasing ODBC
> handle 0x7f8da8002938 into pool
> [Oct 12 16:45:24] VERBOSE[26541][C-000e] pbx_realtime.c: Executing
> [5206700792 <(520)%20670-0792>@sip-in:6] Goto("PJSIP/trunk1-000e",
> "kiniston-ivr,s,1")
> [Oct 12 16:45:24] VERBOSE[26541][C-000e] pbx_builtins.c: Goto
> (kiniston-ivr,s,1)
>
>
>
> The contents of my table are just
> idcontextextenpriorityappappdata
> 7301702413361087843 sip-in5206700792 <(520)%20670-0792>5
> Gosubsub-setupinboundcall,s,1(kiniston,Pop_Country_Crossover,300,VOIP)
> 7301702413361087848sip-in5206700792 <(520)%20670-0792>6
> Gotokiniston-ivr,s,1
>
> My extensions.conf
> [sip-in]
> exten => _X.,1,Log(NOTICE,Incoming ${CHANNEL:0:3} call DID: ${EXTEN} from
> CallerID: ${CALLERID(num)} ${CALLERID(name)}, ANI:${CALLERID(ani)}
> ANI2:${CALLINGANI2})
> exten => _X.,2,Set(ARRAY(CDR(firstext),CDR(firstcontext))=${EXTEN},${
> CONTEXT})
> exten => _X.,3,Set(ARRAY(__FirstEXT,__FirstContext)=${EXTEN},${CONTEXT})
> exten => _X.,4,Set(CHANNEL(hangup_handler_push)=cdr-fixup,s,1)
> exten => _X.,7,Hangup()
> switch => Realtime/sip-in@extensions/p
>
>
> Has anyone else noticed this behavior? Is it expected? I dont' see any
> mentions of it in any the docs.
>
> --
> 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, 

[asterisk-users] Realtime extensions, Multiple SQL lookups?

[John Kiniston]

I'm toying with the idea of replacing a statically generated file I include
in my extensions.conf with a realtime lookup against my database.

I've got it working but something seems off in my logs, It looks like I'm
getting two lookups for every priority?

[Oct 12 16:45:24] DEBUG[26541][C-000e] res_odbc.c: Reusing ODBC handle
0x7f8da8002938 from class 'odbc_kiniston-test'
[Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Skip: 0; SQL:
SELECT * FROM extensions-test WHERE exten = ? AND priority = ? AND context
= ?
[Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Parameter 1
('exten') = '5206700792'
[Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Parameter 2
('priority') = '5'
[Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Parameter 3
('context') = 'sip-in'
[Oct 12 16:45:24] DEBUG[26541][C-000e] res_odbc.c: Releasing ODBC
handle 0x7f8da8002938 into pool
[Oct 12 16:45:24] DEBUG[26541][C-000e] res_odbc.c: Reusing ODBC handle
0x7f8da8002938 from class 'odbc_kiniston-test'
[Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Skip: 0; SQL:
SELECT * FROM extensions-test WHERE exten = ? AND priority = ? AND context
= ?
[Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Parameter 1
('exten') = '5206700792'
[Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Parameter 2
('priority') = '5'
[Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Parameter 3
('context') = 'sip-in'
[Oct 12 16:45:24] DEBUG[26541][C-000e] res_odbc.c: Releasing ODBC
handle 0x7f8da8002938 into pool
[Oct 12 16:45:24] VERBOSE[26541][C-000e] pbx_realtime.c: Executing
[5206700792@sip-in:5] Gosub("PJSIP/trunks1-000e",
"sub-setupinboundcall,s,1(kiniston,Pop_Country_Crossover,300,VOIP)")

---SNIP---

[Oct 12 16:45:24] DEBUG[26541][C-000e] res_odbc.c: Reusing ODBC handle
0x7f8da8002938 from class 'odbc_kiniston-test'
[Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Skip: 0; SQL:
SELECT * FROM extensions-test WHERE exten = ? AND priority = ? AND context
= ?
[Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Parameter 1
('exten') = '5206700792'
[Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Parameter 2
('priority') = '6'
[Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Parameter 3
('context') = 'sip-in'
[Oct 12 16:45:24] DEBUG[26541][C-000e] res_odbc.c: Releasing ODBC
handle 0x7f8da8002938 into pool
[Oct 12 16:45:24] DEBUG[26541][C-000e] res_odbc.c: Reusing ODBC handle
0x7f8da8002938 from class 'odbc_kiniston-test'
[Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Skip: 0; SQL:
SELECT * FROM extensions-test WHERE exten = ? AND priority = ? AND context
= ?
[Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Parameter 1
('exten') = '5206700792'
[Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Parameter 2
('priority') = '6'
[Oct 12 16:45:24] DEBUG[26541][C-000e] res_config_odbc.c: Parameter 3
('context') = 'sip-in'
[Oct 12 16:45:24] DEBUG[26541][C-000e] res_odbc.c: Releasing ODBC
handle 0x7f8da8002938 into pool
[Oct 12 16:45:24] VERBOSE[26541][C-000e] pbx_realtime.c: Executing
[5206700792@sip-in:6] Goto("PJSIP/trunk1-000e", "kiniston-ivr,s,1")
[Oct 12 16:45:24] VERBOSE[26541][C-000e] pbx_builtins.c: Goto
(kiniston-ivr,s,1)



The contents of my table are just
idcontextextenpriorityappappdata
7301702413361087843 sip-in52067007925Gosub
sub-setupinboundcall,s,1(kiniston,Pop_Country_Crossover,300,VOIP)
7301702413361087848sip-in52067007926Gotokiniston-ivr,s,1

My extensions.conf
[sip-in]
exten => _X.,1,Log(NOTICE,Incoming ${CHANNEL:0:3} call DID: ${EXTEN} from
CallerID: ${CALLERID(num)} ${CALLERID(name)}, ANI:${CALLERID(ani)}
ANI2:${CALLINGANI2})
exten =>
_X.,2,Set(ARRAY(CDR(firstext),CDR(firstcontext))=${EXTEN},${CONTEXT})
exten => _X.,3,Set(ARRAY(__FirstEXT,__FirstContext)=${EXTEN},${CONTEXT})
exten => _X.,4,Set(CHANNEL(hangup_handler_push)=cdr-fixup,s,1)
exten => _X.,7,Hangup()
switch => Realtime/sip-in@extensions/p


Has anyone else noticed this behavior? Is it expected? I dont' see any
mentions of it in any the docs.

-- 
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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asterisk-users 

Re: [asterisk-users] Realtime pjsip issues

[Bryant Zimmerman]

 
  Original Message 
> From: "Joshua Colp" <jc...@digium.com>
> Sent: Friday, September 15, 2017 11:31 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Realtime pjsip issues
>
> On Fri, Sep 15, 2017, at 12:18 PM, Bryant Zimmerman wrote:
> > Joshua
> >
> > We are using MariaDB as the database storage.
> > We have recreated the database tables with alembic.
> >
> > Test 1:
> > We enable tables for aors, auths and endpoints only. With cache turned
> > off the end point registers successfully We have no way to get any
> > feed
> > back as pjsip show/list returns no objects found. pjsip send notify
> > cmd
> > endpoint -- does not work as it says there is no endpoint. endpoint
> > can
> > send a call as it appears to be registered, we have no way to confirm
> > this
> > form the console but calls come in.
>
> 
>
> The show and list commands are supposed to work, even without caching
> being enabled. Your problem is therefore at the realtime level. Calls
> coming in should appear on the console, and the endpoint name will be in
> the channel name. Enabling caching just masks it some because things
> exist in the cache for a bit.
>
> >
> > I can offer the following:
> > A dump of the database schema that alembic is creating.
> > extconfig.config
> > sorcery.conf
>
> Feel free to provide these and me (or another individual) may pick out
> what is wrong.
>
> --
> 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
>
  
 I have linked to a zip file containing a dump of my sql schema (MySQL), 
extconfig.conf, sorcery.conf
  
 dumps.zip
  
 Hopefully someone can see what might be causing our issues with the pjsip 
realtime system. 
  
 Thanks 
 Bryant
 



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Re: [asterisk-users] Realtime pjsip issues

[Joshua Colp]

On Fri, Sep 15, 2017, at 12:18 PM, Bryant Zimmerman wrote:
> Joshua
>   
>  We are using MariaDB as the database storage. 
>  We have recreated the database tables with alembic. 
>   
>  Test 1:
>   We enable tables for aors, auths and endpoints only.With cache 
> turned 
> off the end point registers successfullyWe have no way to get any
> feed 
> back as pjsip show/list returns no objects found.   pjsip send notify
> cmd 
> endpoint -- does not work as it says there is no endpoint.  endpoint
> can 
> send a call as it appears to be registered, we have no way to confirm
> this 
> form the console but calls come in.  



The show and list commands are supposed to work, even without caching
being enabled. Your problem is therefore at the realtime level. Calls
coming in should appear on the console, and the endpoint name will be in
the channel name. Enabling caching just masks it some because things
exist in the cache for a bit.

>   
>  I can offer the following:
>  A dump of the database schema that alembic is creating.
>  extconfig.config
>  sorcery.conf

Feel free to provide these and me (or another individual) may pick out
what is wrong.

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

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Re: [asterisk-users] Realtime pjsip issues

[Marcelo Terres]

Hello.

Did you ps_contacts table has all columns listed here?

INSERT INTO ps_contacts (id, via_addr, qualify_timeout, call_id,
reg_server, path, endpoint, via_port, authenticate_qualify, uri,
qualify_frequency, user_agent, expiration_time, outbound_proxy) VALUES
(?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?)

Regards,
Marcelo H. Terres <mhter...@gmail.com>
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres


On 15 September 2017 at 16:18, Bryant Zimmerman <brya...@zktech.com> wrote:
> Joshua
>
> We are using MariaDB as the database storage.
> We have recreated the database tables with alembic.
>
> Test 1:
>
> We enable tables for aors, auths and endpoints only.
> With cache turned off the end point registers successfully
> We have no way to get any feed back as pjsip show/list returns no objects
> found.
> pjsip send notify cmd endpoint -- does not work as it says there is no
> endpoint.
> endpoint can send a call as it appears to be registered, we have no way to
> confirm this form the console but calls come in.
>
>
> Test 2:
>
> We enable cache on the endpoints, auth and aors in the sorcery.conf
>
> endpoint/cache =
> memory_cache,object_lifetime_stale=600,object_lifetime_maximum=1800,expire_on_reload=yes,full_backend_cache=yes
> auth/cache=memory_cache,expire_on_reload=yes
> aor/cache =
> memory_cache,object_lifetime_stale=1500,object_lifetime_maximum=1800,expire_on_reload=yes,full_backend_cache=yes
>
> We now get an error:
>
> [2017-09-15 11:02:04] WARNING[3375]: res_pjsip_registrar.c:744
> registrar_on_rx_request: AOR '6162480909-300' has no configured
> max_contacts. Endpoint '6162480909-300' unable to register
> The aors entry has the max_contacts set to 1 but the error still occurs.
>
> pjsip show/list shows the endpoint shows endpoints, aors, auths  but
> registration fails
>
>
> Test 3:
>
> We enable cache on the endpoints, auth and aors in the sorcery.conf
>
> endpoint/cache =
> memory_cache,object_lifetime_stale=600,object_lifetime_maximum=1800,expire_on_reload=yes
> auth/cache=memory_cache,expire_on_reload=yes
> aor/cache =
> memory_cache,object_lifetime_stale=1500,object_lifetime_maximum=1800,expire_on_reload=yes
>
> Endpoint registers
> pjsip show/list endpoints works the first time and fails there after.
>
> UBNTU-ROSSI-GUEST*CLI> pjsip show endpoints
>  Endpoint:  
>   
> I/OAuth:
> 
> Aor:  
> 
>   Contact:   
>  <RTT(ms)..>
>   Transport:
> 
>Identify:
> 
> Match:  
> Channel:  
>   
> Exten:   CLCID: 
> ==
>  Endpoint:  6162480909-300   Not in use
> 0 of inf
>  InAuth:  6162480909-300/6162480909-300
> Aor:  6162480909-300 1
>   Contact:  6162480909-300/sip:6162480909-300@192.168. 0475d46ff2
> Unknown nan
>   Transport:  udp-nat   udp  0  0  0.0.0.0:5060
>
> Objects found: 1
> UBNTU-ROSSI-GUEST*CLI> pjsip show endpoints
> No objects found.
>
> pjsip show/list shows the endpoint fails ever time after the first.
>
> Test 4:
>
> Test 1: with the addition of the contacts entry as realtime in sorcery.conf
> We get error on registration attempt:
>
> [2017-09-15 11:16:07] WARNING[3591]: res_config_odbc.c:120 custom_prepare:
> SQL Prepare failed! [INSERT INTO ps_contacts (id, via_addr, qualify_timeout,
> call_id, reg_server, path, endpoint, via_port, authenticate_qualify, uri,
> qualify_frequency, user_agent, expiration_time, outbound_proxy) VALUES (?,
> ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?)]
> [2017-09-15 11:16:07] ERROR[3591]: res_pjsip_registrar.c:432
> register_aor_core: Unable to bind contact
> 'sip:6162480909-300@192.168.201.105:59758' to AOR '6162480909-300'
>
> Registration has failed at this point.
>
>
> I can offer the following:
> A dump of the database schema that alembic is creating.
> extconfig.config
> sorcery.conf
>
> Thanks
> Bryant
>
> 
> From: "Joshua Colp" <jc...@digium.com>
> Sent: Friday, September 15, 2017 9:56 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Realtime pjsip issues
>
> On Fri, Sep 15, 2017, at 10:37 AM, Bryant Zimmerman wrote:
>> Joshua
>>
>> That is the interesting part of it. We took our configs and database
>> tables from our working 13.12.2 deployments and tried to use them with
>> our
>> new 13.17.1 deployments and we ar

Re: [asterisk-users] Realtime pjsip issues

[Bryant Zimmerman]

Joshua
  
 We are using MariaDB as the database storage. 
 We have recreated the database tables with alembic. 
  
 Test 1:
We enable tables for aors, auths and endpoints only.With cache 
turned 
off the end point registers successfullyWe have no way to get any feed 
back as pjsip show/list returns no objects found.   pjsip send notify cmd 
endpoint -- does not work as it says there is no endpoint.  endpoint can 
send a call as it appears to be registered, we have no way to confirm this 
form the console but calls come in.  
  
 Test 2: 
We enable cache on the endpoints, auth and aors in the sorcery.conf 
 
endpoint/cache = 
memory_cache,object_lifetime_stale=600,object_lifetime_maximum=1800,expire_o
n_reload=yes,full_backend_cache=yes 
auth/cache=memory_cache,expire_on_reload=yes
aor/cache = 
memory_cache,object_lifetime_stale=1500,object_lifetime_maximum=1800,expire_
on_reload=yes,full_backend_cache=yes
We now get an error:[2017-09-15 
11:02:04] WARNING[3375]: 
res_pjsip_registrar.c:744 registrar_on_rx_request: AOR '6162480909-300' has 
no configured max_contacts. Endpoint '6162480909-300' unable to register 
The aors entry has the max_contacts set to 1 but the error 
still occurs.  

pjsip show/list shows the endpoint shows endpoints, aors, 
auths  but 
registration fails 
  
  Test 3: 
We enable cache on the endpoints, auth and aors in the sorcery.conf 
 
endpoint/cache = 
memory_cache,object_lifetime_stale=600,object_lifetime_maximum=1800,expire_o
n_reload=yesauth/cache=memory_cache,expire_on_reload=yes
aor/cache = 
memory_cache,object_lifetime_stale=1500,object_lifetime_maximum=1800,expire_
on_reload=yes   
Endpoint registers  pjsip show/list endpoints works the 
first time and 
fails there after.  UBNTU-ROSSI-GUEST*CLI> pjsip 
show endpoints
 Endpoint:
  
I/OAuth:  


Aor:

  Contact:
 <RTT(ms)..>
  Transport:  

   Identify:  


Match:  
Channel:
  
Exten:   CLCID: 


===
===
 Endpoint:  6162480909-300   Not in 
use0 of inf
 InAuth:  6162480909-300/6162480909-300
Aor:  6162480909-300 1
  Contact:  6162480909-300/sip:6162480909-300@192.168. 0475d46ff2 
Unknown nan
  Transport:  udp-nat   udp  0  0  0.0.0.0:5060

Objects found: 1
UBNTU-ROSSI-GUEST*CLI> pjsip show endpoints
No objects found.

pjsip show/list shows the endpoint fails ever time after the 
first. 

 Test 4: 
Test 1: with the addition of the contacts entry as realtime in 
sorcery.confWe get error on registration attempt:   
[2017-09-15 
11:16:07] WARNING[3591]: res_config_odbc.c:120 custom_prepare: SQL Prepare 
failed! [INSERT INTO ps_contacts (id, via_addr, qualify_timeout, call_id, 
reg_server, path, endpoint, via_port, authenticate_qualify, uri, 
qualify_frequency, user_agent, expiration_time, outbound_proxy) VALUES (?, 
?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?)] [2017-09-15 11:16:07] 
ERROR[3591]: res_pjsip_registrar.c:432 register_aor_core: Unable to bind 
contact 'sip:6162480909-300@192.168.201.105:59758' to AOR '6162480909-300' 

Registration has failed at this point.  
  
 I can offer the following:
 A dump of the database schema that alembic is creating.
 extconfig.config
 sorcery.conf
  
 Thanks
 Bryant
  


 From: "Joshua Colp" <jc...@digium.com>
Sent: Friday, September 15, 2017 9:56 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Realtime pjsip issues   
On Fri, Sep 15, 2017, at 10:37 AM, Bryant Zimmerman wrote:
> Joshua
>
> That is the interesting part of it. We took our configs and database
> tables from our working 13.12.2 deployments and tried to use them with
> our
> new 13.17.1 deployments and we are having issues where the tables are 
not
> working. On the new server asterisk keeps saying it can't find the AORS
> entries when we purge the sorcery memory cache it starts finding the 
aors
> but then it says it cant find the auths.
>
> The wired thing is when it says it can't find the aors and auths entries
> it does not show it is looking for the values in the aors and auth 
fields
> from the endpoints tables. It keeps putting the value from the endpoints
> id
> field as the entries it can't find.
>
> One point of

Re: [asterisk-users] Realtime pjsip issues

[Ryan, Travis]

Any weirdness with realtime has almost always gone back to schema issue for me. 
Just my experience…


On 9/15/17, 10:48 AM, "asterisk-users-boun...@lists.digium.com on behalf of 
Joshua Colp"  wrote:

On Fri, Sep 15, 2017, at 11:38 AM, Bryant Zimmerman wrote:
> Joshua
>   
>  We have completed more testing this morning and when we remove the 
> realtime cache options from the sorcery file the endpoints complete 
> registration, but we pjsip show/list does not offer any feed back at all, 
> We also can't send any pjsip send notify commands as they say they don't 
> have an endpoint there. Something has changed in the cache part of the 
> system that is breaking the system in some manner for us with the current 
> version and we are out of ideas. 

You're still confusing me here. If you've removed the cache, then it's
not being used anymore so I don't see how it can be a problem. If the
commands aren't listing things when you have no cache even in use then
that would point to realtime, not the cache. You'd need to do as I said
with debug to see what queries are being done to confirm things. You
need to do troubleshooting and isolate things to determine the cause of
the problem. You also did not answer my questions about the database
schema.

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

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Re: [asterisk-users] Realtime pjsip issues

[Joshua Colp]

On Fri, Sep 15, 2017, at 11:38 AM, Bryant Zimmerman wrote:
> Joshua
>   
>  We have completed more testing this morning and when we remove the 
> realtime cache options from the sorcery file the endpoints complete 
> registration, but we pjsip show/list does not offer any feed back at all, 
> We also can't send any pjsip send notify commands as they say they don't 
> have an endpoint there. Something has changed in the cache part of the 
> system that is breaking the system in some manner for us with the current 
> version and we are out of ideas. 

You're still confusing me here. If you've removed the cache, then it's
not being used anymore so I don't see how it can be a problem. If the
commands aren't listing things when you have no cache even in use then
that would point to realtime, not the cache. You'd need to do as I said
with debug to see what queries are being done to confirm things. You
need to do troubleshooting and isolate things to determine the cause of
the problem. You also did not answer my questions about the database
schema.

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

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Re: [asterisk-users] Realtime pjsip issues

[Bryant Zimmerman]

Joshua
  
 We have completed more testing this morning and when we remove the 
realtime cache options from the sorcery file the endpoints complete 
registration, but we pjsip show/list does not offer any feed back at all, 
We also can't send any pjsip send notify commands as they say they don't 
have an endpoint there. Something has changed in the cache part of the 
system that is breaking the system in some manner for us with the current 
version and we are out of ideas. 
  
 Thanks
 Bryant
  
  
  
  
  Joshua
  
 That is the interesting part of it. We took our configs and database 
tables from our working 13.12.2 deployments and tried to use them with our 
new 13.17.1 deployments and we are having issues where the tables are not 
working. On the new server asterisk keeps saying it can't find the AORS 
entries when we purge the sorcery memory cache it starts finding the aors 
but then it says it cant find the auths. 
  
 The wired thing is when it says it can't find the aors and auths entries 
it does not show it is looking for the values in the aors and auth fields 
from the endpoints tables. It keeps putting the value from the endpoints id 
field as the entries it can't find. 
  
 One point of note the tables we used and created for pjsip back when we 
setup the 13.12.2 version are not what is currently being created when we 
run alembic now.. Also the contact table from alembic creation process does 
not work we get insert errors inside of asterisk when contact entry 
attempts are being crated. It shows a number of fields that are not there 
in the created tables. 
  
 This is the foundation of my issues. I really have to resolve them in some 
manner so I can mover forward with getting these new systems into 
production. 
 Any assistance is appreciated. 
  
 Thanks
 Bryant
  


 From: "Joshua Colp" <jc...@digium.com>
Sent: Thursday, September 14, 2017 4:34 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Realtime pjsip issues
 On Thu, Sep 14, 2017, at 05:27 PM, Bryant Zimmerman wrote:
> This appears to be some kind of cache issue.
> We have been doing caching with earlier versions of asterisk 13 on the
> pjsip realtime, but now for some reason
> The items only show up the first time we use pjsip list/show and then
> they
> are wiped. I see a new full cache option and that appears to make a
> difference, but it is unclear what is going on. In effect it appears 
that
> items loaded from a database for pjsip must be fully cached or you can't
> look up any data.
>
> Why has a change of this magnitude been put into an LTS?
> What is the best practices. I see in some of the wikis cache
> suggestions.
> What are others really seeing?

There haven't been any changes made except for bug fixes to the sorcery
memory cache, certainly no behavior changes. In fact the implementation
is the same between 13 and 14 except for a single line addition. What is
your sorcery.conf for both?

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

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Re: [asterisk-users] Realtime pjsip issues

[Joshua Colp]

On Fri, Sep 15, 2017, at 10:37 AM, Bryant Zimmerman wrote:
> Joshua
>   
>  That is the interesting part of it. We took our configs and database 
> tables from our working 13.12.2 deployments and tried to use them with
> our 
> new 13.17.1 deployments and we are having issues where the tables are not 
> working. On the new server asterisk keeps saying it can't find the AORS 
> entries when we purge the sorcery memory cache it starts finding the aors 
> but then it says it cant find the auths. 
>   
>  The wired thing is when it says it can't find the aors and auths entries 
> it does not show it is looking for the values in the aors and auth fields 
> from the endpoints tables. It keeps putting the value from the endpoints
> id 
> field as the entries it can't find. 
>   
>  One point of note the tables we used and created for pjsip back when we 
> setup the 13.12.2 version are not what is currently being created when we 
> run alembic now.. Also the contact table from alembic creation process
> does 
> not work we get insert errors inside of asterisk when contact entry 
> attempts are being crated. It shows a number of fields that are not there 
> in the created tables. 
>   
>  This is the foundation of my issues. I really have to resolve them in
>  some 
> manner so I can mover forward with getting these new systems into 
> production. 
>  Any assistance is appreciated. 

You're really throwing a lot of things in here. Please try to simplify
this first and remove the caching. After that what exactly does it say
it can't find when trying to add a contact? What database is in use? Did
you create the tables fresh from alembic? Upgrade existing? If you
enable debug (debug to console in logger.conf and core set debug 5) does
it show it doing database queries?

As it is right now things haven't been narrowed down enough for me to
give any concrete answer or help.

-- 
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445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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Re: [asterisk-users] Realtime pjsip issues

[Bryant Zimmerman]

Joshua
  
 That is the interesting part of it. We took our configs and database 
tables from our working 13.12.2 deployments and tried to use them with our 
new 13.17.1 deployments and we are having issues where the tables are not 
working. On the new server asterisk keeps saying it can't find the AORS 
entries when we purge the sorcery memory cache it starts finding the aors 
but then it says it cant find the auths. 
  
 The wired thing is when it says it can't find the aors and auths entries 
it does not show it is looking for the values in the aors and auth fields 
from the endpoints tables. It keeps putting the value from the endpoints id 
field as the entries it can't find. 
  
 One point of note the tables we used and created for pjsip back when we 
setup the 13.12.2 version are not what is currently being created when we 
run alembic now.. Also the contact table from alembic creation process does 
not work we get insert errors inside of asterisk when contact entry 
attempts are being crated. It shows a number of fields that are not there 
in the created tables. 
  
 This is the foundation of my issues. I really have to resolve them in some 
manner so I can mover forward with getting these new systems into 
production. 
 Any assistance is appreciated. 
  
 Thanks
 Bryant
  


 From: "Joshua Colp" <jc...@digium.com>
Sent: Thursday, September 14, 2017 4:34 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Realtime pjsip issues   
On Thu, Sep 14, 2017, at 05:27 PM, Bryant Zimmerman wrote:
> This appears to be some kind of cache issue.
> We have been doing caching with earlier versions of asterisk 13 on the
> pjsip realtime, but now for some reason
> The items only show up the first time we use pjsip list/show and then
> they
> are wiped. I see a new full cache option and that appears to make a
> difference, but it is unclear what is going on. In effect it appears 
that
> items loaded from a database for pjsip must be fully cached or you can't
> look up any data.
>
> Why has a change of this magnitude been put into an LTS?
> What is the best practices. I see in some of the wikis cache
> suggestions.
> What are others really seeing?

There haven't been any changes made except for bug fixes to the sorcery
memory cache, certainly no behavior changes. In fact the implementation
is the same between 13 and 14 except for a single line addition. What is
your sorcery.conf for both?

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

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Re: [asterisk-users] Realtime pjsip issues

[Joshua Colp]

On Thu, Sep 14, 2017, at 05:27 PM, Bryant Zimmerman wrote:
> This appears to be some kind of cache issue. 
>  We have been doing caching with earlier versions of asterisk 13 on the 
> pjsip realtime, but now for some reason
>  The items only show up the first time we use pjsip list/show and then
>  they 
> are wiped. I see a new full cache option and that appears to make a 
> difference, but it is unclear what is going on. In effect it appears that 
> items loaded from a database for pjsip must be fully cached or you can't 
> look up any data. 
>   
>  Why has a change of this magnitude been put into an LTS?
>  What is the best practices. I see in some of the wikis cache
>  suggestions. 
> What are others really seeing?

There haven't been any changes made except for bug fixes to the sorcery
memory cache, certainly no behavior changes. In fact the implementation
is the same between 13 and 14 except for a single line addition. What is
your sorcery.conf for both?

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

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Re: [asterisk-users] Realtime pjsip issues

[Bryant Zimmerman]

This appears to be some kind of cache issue. 
 We have been doing caching with earlier versions of asterisk 13 on the 
pjsip realtime, but now for some reason
 The items only show up the first time we use pjsip list/show and then they 
are wiped. I see a new full cache option and that appears to make a 
difference, but it is unclear what is going on. In effect it appears that 
items loaded from a database for pjsip must be fully cached or you can't 
look up any data. 
  
 Why has a change of this magnitude been put into an LTS?
 What is the best practices. I see in some of the wikis cache suggestions. 
What are others really seeing?

Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
  


 From: "Bryant Zimmerman" <brya...@zktech.com>
Sent: Thursday, September 14, 2017 2:43 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Realtime pjsip issues   
 We are having an issue where on the latest version of asterisk when 
configuration pjsip via realtime. 

   we do a pjsip list endpoints  it shows our endpoints but lists them as 
invalid. 
   When we do the pjsip list endpoints again it shows no objects. 

   This applies to pjsip list aors as well.  We did not have this issue on 
our older asterisk 13 installs. My guess is something has changed with 
pjsip and realtime. Anyone have any ideas where I can start. We have tried 
a number of things already and would love some suggestions. 


Thanks
zktech


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[asterisk-users] Realtime pjsip issues

[Bryant Zimmerman]

We are having an issue where on the latest version of asterisk when 
configuration pjsip via realtime. 
  
 we do a pjsip list endpoints  it shows our endpoints but lists them as 
invalid. 
 When we do the pjsip list endpoints again it shows no objects. 
  
 This applies to pjsip list aors as well.  We did not have this issue on 
our older asterisk 13 installs. My guess is something has changed with 
pjsip and realtime. Anyone have any ideas where I can start. We have tried 
a number of things already and would love some suggestions. 
  

Thanks
zktech

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Re: [asterisk-users] Realtime queue & agent groups

[Jonas Kellens]

Hello

any one have some input on this ?

I've already tried changing the membername to : testacc77000/@1
Is completely ignored.

I've already tried changing the interface to : testacc77000/@1
Is completely ignored.



Or is it just not possible to group queue members ??



Thanks.

J.


On 27-10-16 15:53, Jonas Kellens wrote:

Hello

I'm a bit confused on how to group agents (give agents a group number) 
when using realtime queues.


I read on the wiki :

  * If you include groups in your queue definition the calls get
routed in the order of the group regardless of the specified
strategy. So I just have a member= line for each agent.

member => Agent/@1 ; a group
member => Agent/501 ; a single agent
member => Agent/:1,1 ; Any agent in group 1, wait for first available, 
but consider with penalty



In my realtime database I have table queue_members :

+--++-++-+-++
| uniqueid | membername | queue_name  | 
interface  | state_interface | penalty 
| paused |

+--++-++-+-++
| 2916 | testacc77000   | queue7700q4 | testacc77000 
| |   0 |   NULL |
| 2917 | testacc77001   | queue7700q4 | testacc77001 
| |   3 |   NULL |
| 2843 | testacc77000   | queue7700q4 | testacc77000 
| |   0 |   NULL |
| 2905 | testacc7700905 | queue7700q5 | testacc7700905 
| |   0 |   NULL |
| 2888 | testacc77000   | queue7700q5 | testacc77000 
| |   0 |   NULL |
| 2900 | testacc77000   | queue7700q5 | testacc77000 
| |   0 |   NULL |
| 2901 | testacc77001   | queue7700q5 | testacc77001 
| |   0 |   NULL |




How do I define a group to a certain agent/member in this case ?





Kind regards

J.






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[asterisk-users] Realtime queue & agent groups

[Jonas Kellens]

Hello

I'm a bit confused on how to group agents (give agents a group number) 
when using realtime queues.


I read on the wiki :

 * If you include groups in your queue definition the calls get routed
   in the order of the group regardless of the specified strategy. So I
   just have a member= line for each agent.

member => Agent/@1 ; a group
member => Agent/501 ; a single agent
member => Agent/:1,1 ; Any agent in group 1, wait for first available, 
but consider with penalty



In my realtime database I have table queue_members :

+--++-++-+-++
| uniqueid | membername | queue_name  | 
interface  | state_interface | penalty | 
paused |

+--++-++-+-++
| 2916 | testacc77000   | queue7700q4 | testacc77000 
| |   0 |   NULL |
| 2917 | testacc77001   | queue7700q4 | testacc77001 
| |   3 |   NULL |
| 2843 | testacc77000   | queue7700q4 | testacc77000 
| |   0 |   NULL |
| 2905 | testacc7700905 | queue7700q5 | testacc7700905 
| |   0 |   NULL |
| 2888 | testacc77000   | queue7700q5 | testacc77000 
| |   0 |   NULL |
| 2900 | testacc77000   | queue7700q5 | testacc77000 
| |   0 |   NULL |
| 2901 | testacc77001   | queue7700q5 | testacc77001 
| |   0 |   NULL |




How do I define a group to a certain agent/member in this case ?





Kind regards

J.


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Re: [asterisk-users] Realtime SIP peers do not register any more after upgrade to Asterisk 13

[Jonas Kellens]

Remove yourself !

Don't hijack my thread !



On 17-08-16 14:53, Dario Estupinan wrote:

REMOVE ME please.

2016-08-15 15:16 GMT-05:00 Jonas Kellens >:


Hello

after I have upgraded from Asterisk 12 to
asterisk-certified-13.8-cert1 none of my realtime SIP peers (saved
in MySQL DB) register anymore.


[Aug 15 22:03:43] NOTICE[30098]: chan_sip.c:28451
handle_request_register: Registration from
'>' failed for
'78.119.140.190:5076 ' - Wrong password
[Aug 15 22:04:13] NOTICE[30098]: chan_sip.c:28451
handle_request_register: Registration from
'>' failed for
'78.119.140.190:5072 ' - Wrong password
[Aug 15 22:04:43] NOTICE[30098]: chan_sip.c:28451
handle_request_register: Registration from
'>' failed for
'78.119.140.190:5062 ' - Wrong password
[Aug 15 22:04:46] NOTICE[30098]: chan_sip.c:28451
handle_request_register: Registration from
'>' failed for
'78.119.140.190:5060 ' - Wrong password
[Aug 15 22:04:53] NOTICE[30098]: chan_sip.c:28451
handle_request_register: Registration from
'>' failed for
'78.119.140.190:5060 ' - Wrong password


Is this a known problem ??


Second question I have : can I get the complete list of columns
that can be used in realtime database for sip peers somewhere
(update for Ast 13) ? Are columns like dtlsenable, dtlsverify,
dtlscertfile, dtlscafile, dtlssetup possible ??




Thanks for the help.


Kind regards.

Jonas.

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Re: [asterisk-users] Realtime SIP peers do not register any more after upgrade to Asterisk 13

[John Novack]

Remove yourself

READ - Included with every message -

asterisk-users mailing list
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Dario Estupinan wrote:


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Re: [asterisk-users] Realtime SIP peers do not register any more after upgrade to Asterisk 13

[Dario Estupinan]

REMOVE ME please.

2016-08-15 15:16 GMT-05:00 Jonas Kellens :

> Hello
>
> after I have upgraded from Asterisk 12 to asterisk-certified-13.8-cert1
> none of my realtime SIP peers (saved in MySQL DB) register anymore.
>
>
> [Aug 15 22:03:43] NOTICE[30098]: chan_sip.c:28451 handle_request_register:
> Registration from '' failed for '
> 78.119.140.190:5076' - Wrong password
> [Aug 15 22:04:13] NOTICE[30098]: chan_sip.c:28451 handle_request_register:
> Registration from '' failed for '
> 78.119.140.190:5072' - Wrong password
> [Aug 15 22:04:43] NOTICE[30098]: chan_sip.c:28451 handle_request_register:
> Registration from '' failed for '
> 78.119.140.190:5062' - Wrong password
> [Aug 15 22:04:46] NOTICE[30098]: chan_sip.c:28451 handle_request_register:
> Registration from '' failed for '
> 78.119.140.190:5060' - Wrong password
> [Aug 15 22:04:53] NOTICE[30098]: chan_sip.c:28451 handle_request_register:
> Registration from '' failed for '
> 78.119.140.190:5060' - Wrong password
>
>
> Is this a known problem ??
>
>
> Second question I have : can I get the complete list of columns that can
> be used in realtime database for sip peers somewhere (update for Ast 13) ?
> Are columns like dtlsenable, dtlsverify, dtlscertfile, dtlscafile,
> dtlssetup possible ??
>
>
>
>
> Thanks for the help.
>
>
> Kind regards.
>
> Jonas.
>
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> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



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Antes de imprimir este mensaje, asegúrese de que es necesario. Proteger el
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a su remitente. Está prohibida su retención, grabación, utilización,
aprovechamiento o divulgación con cualquier propósito. La Corporación
Politécnica Nacional de Colombia no asume ninguna responsabilidad por
eventuales daños generados por el recibo y el uso de este material, siendo
responsabilidad del destinatario verificar con sus propios medios la
existencia de virus u otros defectos. El presente correo electrónico solo
refleja la opinión de su Remitente y no representa necesariamente la
opinión oficial de la Corporación o de sus Directivos.
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Re: [asterisk-users] Realtime SIP peers do not register any more after upgrade to Asterisk 13

[Jonas Kellens]

On 15-08-16 23:00, Carlos Chavez wrote:



I highly recommend that you use alembic to set up your database as 
this will make sure you are always using the correct database schema.  
You should be able to find the "official" structure in the 
contrib/realtime/mysql directory of the Asterisk source.




Hello

in contrib/realtime/mysql I see a table 'sippeers' with a column 
"transport ENUM('udp','tcp','tls','ws','wss','udp,tcp','tcp,udp') " but 
I see no columns dtlsenable, dtlsverify, dtlscertfile, dtlscafile, 
dtlssetup ?


So if we can define a sip peer with transport 'ws' or 'wss', then why 
are there no columns for the 'dtls'-part (which is kinda mandatory) ?




Kind regards.



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Re: [asterisk-users] Realtime SIP peers do not register any more after upgrade to Asterisk 13

[Carlos Chavez]

On 8/15/16 3:16 PM, Jonas Kellens wrote:


Hello

after I have upgraded from Asterisk 12 to 
asterisk-certified-13.8-cert1 none of my realtime SIP peers (saved in 
MySQL DB) register anymore.



[Aug 15 22:03:43] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5076' - 
Wrong password
[Aug 15 22:04:13] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5072' - 
Wrong password
[Aug 15 22:04:43] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5062' - 
Wrong password
[Aug 15 22:04:46] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5060' - 
Wrong password
[Aug 15 22:04:53] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5060' - 
Wrong password



Is this a known problem ??


Second question I have : can I get the complete list of columns that 
can be used in realtime database for sip peers somewhere (update for 
Ast 13) ? Are columns like dtlsenable, dtlsverify, dtlscertfile, 
dtlscafile, dtlssetup possible ??



The first thing you need to test is if you are properly loading the 
realtime data.  The best way would be to enable "rtcachefriends=yes" and 
then "sip show peer XXX load".  If you are not getting anything then 
there is a problem with your realtime setup.  I used realtime sip until 
13.7 before switching to PJSIP so it should work.


I highly recommend that you use alembic to set up your database as 
this will make sure you are always using the correct database schema.  
You should be able to find the "official" structure in the 
contrib/realtime/mysql directory of the Asterisk source.


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Carlos Chávez
+52 (55)9116-91161


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[asterisk-users] Realtime SIP peers do not register any more after upgrade to Asterisk 13

[Jonas Kellens]

Hello

after I have upgraded from Asterisk 12 to asterisk-certified-13.8-cert1 
none of my realtime SIP peers (saved in MySQL DB) register anymore.



[Aug 15 22:03:43] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5076' - Wrong 
password
[Aug 15 22:04:13] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5072' - Wrong 
password
[Aug 15 22:04:43] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5062' - Wrong 
password
[Aug 15 22:04:46] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5060' - Wrong 
password
[Aug 15 22:04:53] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5060' - Wrong 
password



Is this a known problem ??


Second question I have : can I get the complete list of columns that can 
be used in realtime database for sip peers somewhere (update for Ast 13) 
? Are columns like dtlsenable, dtlsverify, dtlscertfile, dtlscafile, 
dtlssetup possible ??





Thanks for the help.


Kind regards.

Jonas.

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[asterisk-users] Realtime warnings for database structure

[Carlos Chavez]

I keep getting messages like these in the cli:

[Aug 10 12:20:17] WARNING[23411]: res_config_mysql.c:1162 require_mysql: 
Realtime table general@ps_contacts: column 'qualify_timeout' cannot be 
type 'int(10)' (need char)
[Aug 10 12:20:17] WARNING[23411]: res_config_mysql.c:1246 require_mysql: 
Possibly unsupported column type 'enum('yes','no')' on column 
'authenticate_qualify'
[Aug 10 12:20:17] WARNING[23411]: res_config_mysql.c:1162 require_mysql: 
Realtime table general@ps_contacts: column 'via_port' cannot be type 
'int(11)' (need char)


Since I am using alembic with the "official" database table 
structures I simply do not understand why.  Who is wrong here, the 
alembic structure or realtime for expecting different things? Obviously 
I do not want to make changes to the database as that will break the 
updates to new versions so it probably needs to be fixed on the realtime 
side.  I just hope that, since they are warnings, they do not affect 
regular operations.


I also get this error every time an endpoint registers to my Asterisk:

[Aug 10 12:24:32] ERROR[23411]: res_pjsip_registrar.c:411 
register_aor_core: Unable to bind contact 
'sip:4...@xx.xx.xx.xx:48007;transport=UDP;rinstance=34b3595c6901f19e' to 
AOR '4001'


I do not know if this is related to the same database problems, I 
have never seen that message when using text files for configuration.  
Contacts do get created:


  Contact:
 

==

  Contact:  4001/sip:4...@xx.xx.xx.xx:48007;transport=UD 653ab7af98 
Created   0.000


ODBC for realtime is still very unstable so I hope we can get a 
little more stability from Mysql while development continues.  So far my 
Asterisk 13.10 installations work fine but the warnings and error clog 
the log files and cli.  I can ignore that but when my clients peer over 
my shoulder they freak out.  Should I open an issue on jira?


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+52 (55)9116-91161


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Re: [asterisk-users] Realtime for PJSIP registrations

[Harley Peters]

On 06/01/2016 12:40 PM, Joshua Colp wrote:

Carlos Chavez wrote:

 I use realtime for my Asterisk configuration and are now making the
transition to Asterisk 13 and PJSIP.  I used alchemy to set up my
databases and I can now configure my endpoints.  While trying to
configure a trunk I can see that there is a database table called
ps_registrations that should be used to make the registration to the
provider but there is no corresponding entry in the sorcery.conf file so
the information is never read into Asterisk.

 Why is this so?  Why put the table there is you cannot use it
(along with the transport table I guess).  Is there a way to activate it
via sorcery.conf?  What would that line look like because just putting
something like "registration=realtime,ps_registrations" in the res_pjsip
section prevents pjsip from loading.


What does it say? The code currently allows this, but you still need to
issue reloads to update things (if you add/change/delete outbound
registrations).



 I tried putting the registration section in the pjsip.conf file but
I am getting an error back from the provider (Fatal response 403).  I
think I am doing everything correctly but I do not know if it is failing
because some of the configuration is in realtime and only the
registration is in the text file.


This sounds like a configuration issue with the outbound registration or
authentication.



 Any advice?  Is realtime ready for production use for PJSIP?


People seem to be using it. Due to some recent changes in how ODBC
support works (we gave more responsibility to UnixODBC for things)
though there have been some crashes and problems which are being
investigated.


[res_pjsip_outbound_registration]
registration=realtime,ps_registrations

This works for me.

Harley

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Re: [asterisk-users] Realtime for PJSIP registrations

[Joshua Colp]

Carlos Chavez wrote:

 I use realtime for my Asterisk configuration and are now making the
transition to Asterisk 13 and PJSIP.  I used alchemy to set up my
databases and I can now configure my endpoints.  While trying to
configure a trunk I can see that there is a database table called
ps_registrations that should be used to make the registration to the
provider but there is no corresponding entry in the sorcery.conf file so
the information is never read into Asterisk.

 Why is this so?  Why put the table there is you cannot use it
(along with the transport table I guess).  Is there a way to activate it
via sorcery.conf?  What would that line look like because just putting
something like "registration=realtime,ps_registrations" in the res_pjsip
section prevents pjsip from loading.


What does it say? The code currently allows this, but you still need to 
issue reloads to update things (if you add/change/delete outbound 
registrations).




 I tried putting the registration section in the pjsip.conf file but
I am getting an error back from the provider (Fatal response 403).  I
think I am doing everything correctly but I do not know if it is failing
because some of the configuration is in realtime and only the
registration is in the text file.


This sounds like a configuration issue with the outbound registration or 
authentication.




 Any advice?  Is realtime ready for production use for PJSIP?


People seem to be using it. Due to some recent changes in how ODBC 
support works (we gave more responsibility to UnixODBC for things) 
though there have been some crashes and problems which are being 
investigated.


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


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[asterisk-users] Realtime for PJSIP registrations

[Carlos Chavez]

I use realtime for my Asterisk configuration and are now making the 
transition to Asterisk 13 and PJSIP.  I used alchemy to set up my 
databases and I can now configure my endpoints.  While trying to 
configure a trunk I can see that there is a database table called 
ps_registrations that should be used to make the registration to the 
provider but there is no corresponding entry in the sorcery.conf file so 
the information is never read into Asterisk.


Why is this so?  Why put the table there is you cannot use it 
(along with the transport table I guess).  Is there a way to activate it 
via sorcery.conf?  What would that line look like because just putting 
something like "registration=realtime,ps_registrations" in the res_pjsip 
section prevents pjsip from loading.


I tried putting the registration section in the pjsip.conf file but 
I am getting an error back from the provider (Fatal response 403).  I 
think I am doing everything correctly but I do not know if it is failing 
because some of the configuration is in realtime and only the 
registration is in the text file.


Any advice?  Is realtime ready for production use for PJSIP?

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Carlos Chávez
+52 (55)9116-91161

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Re: [asterisk-users] Realtime Voicemail MWI

[Stefan Tichy]

>From SIP-debug output I have a little more information.
In one case the first subscribe after a restart was rejected:

SIP/2.0 481 Call/Transaction Does Not Exist


Later subscribes from the same phone needed authentication and after
this where rejected:

SIP/2.0 404 Not found (no mailbox)



There seems to be another problem. Even if the mailbox information
is available, the phones are not notified immediatly when a new
voicemail has arrived. The notification is delayed until the phone
subscribes again.



-- 
Stefan Tichy  ( asterisk3 at pi4tel dot de )

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Re: [asterisk-users] Realtime Voicemail MWI

[Nick Olsen]

I'm bone stock except for the following global(s).
  
 rtcachefriends=yes
limitonpeers=yes
allowsubscribe=yes
notifyringing=yes
notifyhold=yes
notifybusy=yes
  
 My Sip Table looks like (Sorry, Hard to read, Note the 207@103 mailbox 
setting.
  
 90103_14"Nick" <321XXX>103_14   SECRETPASS
103_internalnorfc2833dynamic
port,invite207@103force_rport,comedia   
 yes9001300friendallulaw0yes0   
 2070   
  
  
 And my voicemail table (Hard to read as well)
  
 98207103207PASSNickcentralnoyes
nononono1nonoyesnonoyes
2015-08-17 15:22:09yesno4   
 25
  
  
  
  
 Forcing a reload from the DB with the Prune>Load method loads the mailbox 
about 20% of the time
  
  Named Callgr :
  Nam. Pickupgr:
  MOH Suggest  :
  Mailbox  :
  VM Extension : asterisk
  LastMsgsSent : 1/0
  Call limit   : 0
  Max forwards : 0
  Dynamic  : Yes
  Callerid : "Nick" <321XXX>
  
  
 Nick Olsen
Network Operations  (855) FLSPEED  x106

  


 From: "Stefan Tichy" <asteri...@pi4tel.de>
Sent: Sunday, September 20, 2015 9:28 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Realtime Voicemail MWI   
On Wed, Sep 16, 2015 at 04:44:45PM -0400, Nick Olsen wrote:
> I've got SIP Peers in realtime. All with a mailbox set. 98% of the time,
> These are loaded into asterisk without the mailbox info. Leading to
> "Received SIP subscribe for peer without mailbox" notices. And 
non-working
> MWI.
>
> Occasionally, It just works. But only on a peer or two at a time. And
> it'll stop working after a few minutes.

Here it seems to be the other way round. Occasionally I see that
peers have lost there mailbox setting and don't get notify messages
with voicemail information. It is Asterisk 13.5.0

"sip prune realtime peer ..."
"sip show peer ... load"

After this the setting is restored, but until now I have no idea why
this happens. The database field mailbox remains unchanged.

Could you post the Realtime SIP Settings?

--
Stefan Tichy ( asterisk3 at pi4tel dot de )

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Re: [asterisk-users] Realtime Voicemail MWI

[Stefan Tichy]

On Wed, Sep 16, 2015 at 04:44:45PM -0400, Nick Olsen wrote:
>  I've got SIP Peers in realtime. All with a mailbox set. 98% of the time, 
> These are loaded into asterisk without the mailbox info. Leading to 
> "Received SIP subscribe for peer without mailbox" notices. And non-working 
> MWI.
>   
>  Occasionally, It just works. But only on a peer or two at a time. And 
> it'll stop working after a few minutes.

Here it seems to be the other way round. Occasionally I see that
peers have lost there mailbox setting and don't get notify messages
with voicemail information. It is Asterisk 13.5.0

"sip prune realtime peer ..."
"sip show peer ... load"

After this the setting is restored, but until now I have no idea why
this happens. The database field mailbox remains unchanged.

Could you post the Realtime SIP Settings?
 

-- 
Stefan Tichy  ( asterisk3 at pi4tel dot de )

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Re: [asterisk-users] Realtime Voicemail MWI

[Michele Pinassi]

Hi Nick,

did you set-up also Voicemail boxes in Realtime ?

Michele

Il 16/09/2015 22:44, Nick Olsen ha scritto:
> Greetings All, Regarding this archived
> post. 
> http://lists.digium.com/pipermail/asterisk-users/2014-November/285169.html
>  
> Did anyone ever find an solution to this? I've got a new box running
> 13.3.0 with the exact same issue.
>  
> For those that don't read the link.
>  
> I've got SIP Peers in realtime. All with a mailbox set. 98% of the
> time, These are loaded into asterisk without the mailbox info. Leading
> to "Received SIP subscribe for peer without mailbox" notices. And
> non-working MWI.
>  
> Occasionally, It just works. But only on a peer or two at a time. And
> it'll stop working after a few minutes.
>  
> Any ideas? Thanks
>  
>

-- 
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena
tel: 0577.(23)5000 - fax: 0577.(23)2053

Per trovare una soluzione rapida ai tuoi problemi tecnici
consulta le FAQ di Ateneo, http://www.faq.unisi.it 



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Re: [asterisk-users] Realtime Voicemail MWI

[Nick Olsen]

Yes, They are.

 Nick Olsen
Network Operations  (855) FLSPEED  x106




 From: "Michele Pinassi" <michele.pina...@unisi.it>
Sent: Thursday, September 17, 2015 3:07 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Realtime Voicemail MWI
Hi Nick,

did you set-up also Voicemail boxes in Realtime ?

Michele
  Il 16/09/2015 22:44, Nick Olsen ha scritto:
  Greetings All, Regarding this archived post. 
http://lists.digium.com/pipermail/asterisk-users/2014-November/285169.html

 Did anyone ever find an solution to this? I've got a new box running 13.3.0 
with the exact same issue.

 For those that don't read the link.

 I've got SIP Peers in realtime. All with a mailbox set. 98% of the time, These 
are loaded into asterisk without the mailbox info. Leading to "Received SIP 
subscribe for peer without mailbox" notices. And non-working MWI.

 Occasionally, It just works. But only on a peer or two at a time. And it'll 
stop working after a few minutes.

 Any ideas? Thanks



 --  Michele Pinassi Responsabile Telefonia di Ateneo Servizio Reti, Sistemi e 
Sicurezza Informatica - Università degli Studi di Siena tel: 0577.(23)5000 - 
fax: 0577.(23)2053  Per trovare una soluzione rapida ai tuoi problemi tecnici 
consulta le FAQ di Ateneo, http://www.faq.unisi.it

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[asterisk-users] Realtime Voicemail MWI

[Nick Olsen]

Greetings All, Regarding this archived post. 
http://lists.digium.com/pipermail/asterisk-users/2014-November/285169.html
  
 Did anyone ever find an solution to this? I've got a new box running 
13.3.0 with the exact same issue.
  
 For those that don't read the link.
  
 I've got SIP Peers in realtime. All with a mailbox set. 98% of the time, 
These are loaded into asterisk without the mailbox info. Leading to 
"Received SIP subscribe for peer without mailbox" notices. And non-working 
MWI.
  
 Occasionally, It just works. But only on a peer or two at a time. And 
it'll stop working after a few minutes.
  
 Any ideas? Thanks
  
 Nick Olsen
Network Operations  (855) FLSPEED  x106


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[asterisk-users] Realtime peers and mailbox not existant

[Leandro Dardini]

Some time to time, usually after an asterisk restart or a sip reload, some
realtime sip peers are loaded in memory without their mailbox. I was not
able to replicate the issue on a constant basis, but after adding some
additional logs to asterisk, it seems the add_peer_mailboxes is run
correctly, but then, when the SIP SUBSCRIBE arrives, the mailbox is not
found. If I run a SIP SHOW PEER, the peer is shown without the mailbox.

Have you ever noticed a similar behavior?

Leandro
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[asterisk-users] Realtime peers, mailbox and MWI problem

[Leandro Dardini]

Hello,
I am facing a problem I can't understand. I have several realtime SIP peers
and from time to time, the mailbox field is not loaded in asterisk memory.
The mailbox field is correctly populated in the database, but often, after
an asterisk restart, the mailbox is not associated to the peer (just to
understand, if I run sip show peer 104-TEST, I see the Mailbox empty. If
I run the sip show subscriptiona, I don't see any subscription for the
MWI but only for the BLF.

Is there anyone facing the same problem? How have you solved it?

leandro
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[asterisk-users] Realtime followme and channel variables

[Leandro Dardini]

Followme is perfect to handle FMFM and it is now also realtime, but it
seems impossible to assign some value to a variable, from within the
followme to store info for example about the tenant the followme is running
under, like instead happen for example in the queue with the
setinterfacevar field.

I just need to pass a variable from the channel placing the call to the
followme to the channel where the extension is dialed by followme. Any idea?

Leandro
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Re: [asterisk-users] Realtime followme and channel variables

[Richard Mudgett]

On Thu, Mar 12, 2015 at 5:14 PM, Leandro Dardini ldard...@gmail.com wrote:

 Followme is perfect to handle FMFM and it is now also realtime, but it
 seems impossible to assign some value to a variable, from within the
 followme to store info for example about the tenant the followme is running
 under, like instead happen for example in the queue with the
 setinterfacevar field.

 I just need to pass a variable from the channel placing the call to the
 followme to the channel where the extension is dialed by followme. Any idea?


Sounds like you need to use variable inheritance.

https://wiki.asterisk.org/wiki/display/AST/Variable+Inheritance

Richard
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[asterisk-users] Realtime not storing voicemail password changes

[Paddy Grice]

Hi All
 
I am trying to get voicemail switched over to ARA on version 13 and notice
that the password is not stored in the db when it is changed.
 
A little hair pulling and playing around and I think the problem is in the
function ast_update2_realtime in main/config.c.
 
Issued source is ==
 
int ast_update2_realtime(const char *family, ...)
{
RAII_VAR(struct ast_variable *, lookup_fields, NULL,
ast_variables_destroy);
RAII_VAR(struct ast_variable *, update_fields, NULL,
ast_variables_destroy);
va_list ap;
 
va_start(ap, family);
/* XXX: If we wanted to pass no lookup fields (select all), we'd be
 * out of luck. realtime_arguments_to_fields expects at least one
key
 * value pair. */
realtime_arguments_to_fields(ap, lookup_fields);
va_end(ap);
 
va_start(ap, family);
realtime_arguments_to_fields2(ap, 1, lookup_fields);
va_end(ap);
 
if (!lookup_fields || !update_fields) {
return -1;
}
 
return ast_update2_realtime_fields(family, lookup_fields,
update_fields);
}
 
I believe line 3314 of the file main/config.c should be 
 
realtime_arguments_to_fields2(ap, 1, update_fields);
 
I have changed it and it works for me - but - 
 
1)I don't know what else this may effect 
2)I dont know how to pass this on to the development team
 
Paddy
 
 
 
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Re: [asterisk-users] Realtime not storing voicemail password changes

[Matthew Jordan]

On Tue, Dec 16, 2014 at 6:25 AM, Paddy Grice pa...@wizaner.com wrote:
 Hi All

 I am trying to get voicemail switched over to ARA on version 13 and notice
 that the password is not stored in the db when it is changed.

 A little hair pulling and playing around and I think the problem is in the
 function ast_update2_realtime in main/config.c.

 Issued source is ==

 int ast_update2_realtime(const char *family, ...)
 {
 RAII_VAR(struct ast_variable *, lookup_fields, NULL,
 ast_variables_destroy);
 RAII_VAR(struct ast_variable *, update_fields, NULL,
 ast_variables_destroy);
 va_list ap;

 va_start(ap, family);
 /* XXX: If we wanted to pass no lookup fields (select all), we'd be
  * out of luck. realtime_arguments_to_fields expects at least one
 key
  * value pair. */
 realtime_arguments_to_fields(ap, lookup_fields);
 va_end(ap);

 va_start(ap, family);
 realtime_arguments_to_fields2(ap, 1, lookup_fields);
 va_end(ap);

 if (!lookup_fields || !update_fields) {
 return -1;
 }

 return ast_update2_realtime_fields(family, lookup_fields,
 update_fields);
 }

 I believe line 3314 of the file main/config.c should be

 realtime_arguments_to_fields2(ap, 1, update_fields);

That looks right to me. Otherwise, we never extract the variable list
arguments passed in to the ast_variable list update_fields. Yikes.

 I have changed it and it works for me - but -

 1)I don't know what else this may effect
 2)I dont know how to pass this on to the development team


Please file a bug on issues.asterisk.org.

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com  http://asterisk.org

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Re: [asterisk-users] Realtime ERROR

[Andrew Colin]

Hi Guys,

 

I have recently moved my database servers to a different database cluster
that runs on haproxy.

Every minute or so I get the following error in asterisk

 

MySQL RealTime: Ping failed (2006).  Trying an explicit reconnect

 

The strange thing is if I do realtime mysql status

It shows as connected just the timer resets.

 

Any idea why this is occurring?

 

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Re: [asterisk-users] Realtime ERROR

[Rainer Piper]

Am 25.09.2014 um 16:24 schrieb Andrew Colin:


Hi Guys,

I have recently moved my database servers to a different database 
cluster that runs on haproxy.


Every minute or so I get the following error in asterisk

MySQL RealTime: Ping failed (2006).  Trying an explicit reconnect

The strange thing is if I do realtime mysql status

It shows as connected just the timer resets.

Any idea why this is occurring?




Hi Andrew,

what balancing algorithm you use in haproxy.cfg  ?
balance source
balance roundrobin
or
balance leastconn


--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161 callto:004922897167161
P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test)
XMPP: rai...@xmpp.soho-piper.de
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Re: [asterisk-users] Realtime ERROR

[Andrew Colin]

Hi Rainer,

 

I am using roundrobin

 

From: Rainer Piper [mailto:rainer.pi...@soho-piper.de] 
Sent: Thursday, September 25, 2014 6:21 PM
To: Andrew Colin; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime ERROR

 

Am 25.09.2014 um 16:24 schrieb Andrew Colin:

Hi Guys,

 

I have recently moved my database servers to a different database cluster
that runs on haproxy.

Every minute or so I get the following error in asterisk

 

MySQL RealTime: Ping failed (2006).  Trying an explicit reconnect

 

The strange thing is if I do realtime mysql status

It shows as connected just the timer resets.

 

Any idea why this is occurring?

 





Hi Andrew,

what balancing algorithm you use in haproxy.cfg  ?
balance source
balance roundrobin
or
balance leastconn



-- 
Rainer Piper 
Integration engineer 
Koeslinstr. 56 
53123 BONN 
GERMANY 
Phone:  callto:004922897167161 +49 228 97167161 
P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test) 
XMPP: rai...@xmpp.soho-piper.de

  _  

No virus found in this message.
Checked by AVG - www.avg.com
Version: 2014.0.4765 / Virus Database: 4025/8267 - Release Date: 09/24/14

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Re: [asterisk-users] Realtime integration: Unregistered clients showing as registered?

[Leandro Dardini]

It is the way it works. First the phone sends a REGISTER without any
password. Asterisk answers with a Unauthorized and provide a nonce to be
used for the next registration attempt, using it to encrypt the password.

Leandro


2014-05-14 13:12 GMT+02:00 Olli Heiskanen ohjelmistoarkkite...@gmail.com:


 Hello,

 After a small break from working on this, I got the idea of tcpdumping the
 correct ports. What I see is REGISTER messages from Kamailio port to
 Asterisk, which are replied with 401 Unauthorized. Why is this happening?
 In my sippeers table the secret field has no value (tried both NULL and
 empty string) and the added field sippasswd has the correct password for
 the user.

 The above might be the cause of my problem, would anyone be able to advice
 me to get to correct behaviour? Now Kamailio sees the clients as
 registered, which would be wrong if Asterisk doesn't.

 cheers,
 Olli



 2014-04-24 11:27 GMT+03:00 Olli Heiskanen ohjelmistoarkkite...@gmail.com
 :


 Hello all,

 I've been testing a Kamailio Asterisk Realtime integration, and found a
 strange situation.

 My problem is that when using the integration, everything seems ok but
 Asterisk does not see the clients as registered. Kamailio and the clients
 report registered clients. Also calls fail.

 In Asterisk cli sip show peers shows nothing but for example realtime
 load sipusers name 660 shows the user data. Field regseconds has a value
 and fullcontact has value 'sip:660@127.0.0.1:5060' (kamailio ip:port as
 they are on the same machine).

 I have a very simple dialplan:

 [general]

 [default]
 exten = _XXX,1,NoOp(general : Dialed ${EXTEN})
  same = n,Dial(SIP/${EXTEN},3600,rt)
  same = n,Hangup


 Here's more on my problem and background to it, guys on the Kamailio list
 helped out but looks like I need to check my Asterisk configuration.
 https://www.mail-archive.com/sr-users@lists.sip-router.org/msg18555.html

 My goal is to have all clients in the asterisk database, asterisk (one at
 this point, several later) handling the calls and Kamailio as proxy. In
 Kamailio I have the WITH_MULTIDOMAIN directive on but I'm using only one
 domain 'testers.com'.

 I have Asterisk 11.8.1 and Kamailio 4.2.0-dev4 on CentOS 6.5, all are on
 the same rental virtual server. Clients are in my home network behind nat.
 In MySQL I have database asterisk with table sippeers, where I have
 clients added like this:
 INSERT INTO sippeers
 (name,defaultuser,host,sippasswd,fromuser,fromdomain,callbackextension,type)
 VALUES ('660', '660', 'dynamic', 'password', '660', 'testers.com
 ','660','friend');

 In this message there are some outputs and a sip trace of a register:
 https://www.mail-archive.com/sr-users@lists.sip-router.org/msg18558.html

 What I don't know is how to configure sip.conf, so far I've just been
 making guesses based on online examples and documentation.
 My current sip.conf looks like this:

 [general]
 bindport = 5070
 bindaddr = 127.0.0.1
 tcpbindaddr = 127.0.0.1:5070
 tcpenable = no
 limitonpeers = yes
 ;rtcachefriends = yes
 tos_sip=cs3
 tos_audio=ef
 realm = testers.com

 I've tried defining realm and domain values, but I lack proper
 understanding of those. Can you guys help me out? Are there any other
 configurations I need to check?

 Respectfully,
 Olli




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Re: [asterisk-users] Realtime integration: Unregistered clients showing as registered?

[Olli Heiskanen]

Hello,

Thank you for your response.

Actually, I managed to solve a part of the problem; as I use Kamailio to
handle authentication, problem was that even though authentication went ok
through Kamailio, Asterisk refused to accept messages from Kamailio. That's
why Asterisk sent the 401. I think I had incorrect values in the realtime
sippeers table rows, and also I had to add values to deny and permit
fields, which in fact were in the wrong order. So no wonder I was having
problems with authentication! (and yes, I do know how digest authentication
works ;))

I fixed the deny values to 0.0.0.0/0.0.0.0 and permit value to Kamailio ip.

Even after this I had problems having Asterisk accept the authentications.
Asterisk cli was saying:
ERROR[20605]: chan_sip.c:30790 build_peer: Bad ACL entry in configuration
line 0 : kamailioip:5060

... that was because I had tried to define kamailio ip with port, as
Kamailio and Asterisk are on the same machine, but removing the port solved
that (not sure but I guess it is good I use 5060 for Kamailio and 5070 for
Asterisk instead of vice versa, perhaps this solution wouldn't work then).
Then I found that I had to add values to fields: nat (to force_rport) and
defaultip (to 0.0.0.0), and only after that I got Asterisk to see the
registered peers. So now everything looks ok from both Asterisk and
Kamailio when it comes to authentication.

I still can't get calls going though, in the asterisk cli I get 'Everyone
is busy/congested at this time', so I'm going to continue investigating
that. If you guys have good advice for me at this time I'll be most happy
to take them.

cheers,
Olli



2014-05-15 17:17 GMT+03:00 Leandro Dardini ldard...@gmail.com:

 It is the way it works. First the phone sends a REGISTER without any
 password. Asterisk answers with a Unauthorized and provide a nonce to be
 used for the next registration attempt, using it to encrypt the password.

 Leandro


 2014-05-14 13:12 GMT+02:00 Olli Heiskanen ohjelmistoarkkite...@gmail.com
 :


 Hello,

 After a small break from working on this, I got the idea of tcpdumping
 the correct ports. What I see is REGISTER messages from Kamailio port to
 Asterisk, which are replied with 401 Unauthorized. Why is this happening?
 In my sippeers table the secret field has no value (tried both NULL and
 empty string) and the added field sippasswd has the correct password for
 the user.

 The above might be the cause of my problem, would anyone be able to
 advice me to get to correct behaviour? Now Kamailio sees the clients as
 registered, which would be wrong if Asterisk doesn't.

 cheers,
 Olli



 2014-04-24 11:27 GMT+03:00 Olli Heiskanen ohjelmistoarkkite...@gmail.com
 :


 Hello all,

 I've been testing a Kamailio Asterisk Realtime integration, and found a
 strange situation.

 My problem is that when using the integration, everything seems ok but
 Asterisk does not see the clients as registered. Kamailio and the clients
 report registered clients. Also calls fail.

 In Asterisk cli sip show peers shows nothing but for example realtime
 load sipusers name 660 shows the user data. Field regseconds has a value
 and fullcontact has value 'sip:660@127.0.0.1:5060' (kamailio ip:port as
 they are on the same machine).

 I have a very simple dialplan:

 [general]

 [default]
 exten = _XXX,1,NoOp(general : Dialed ${EXTEN})
  same = n,Dial(SIP/${EXTEN},3600,rt)
  same = n,Hangup


 Here's more on my problem and background to it, guys on the Kamailio
 list helped out but looks like I need to check my Asterisk configuration.
 https://www.mail-archive.com/sr-users@lists.sip-router.org/msg18555.html

 My goal is to have all clients in the asterisk database, asterisk (one
 at this point, several later) handling the calls and Kamailio as proxy. In
 Kamailio I have the WITH_MULTIDOMAIN directive on but I'm using only one
 domain 'testers.com'.

 I have Asterisk 11.8.1 and Kamailio 4.2.0-dev4 on CentOS 6.5, all are on
 the same rental virtual server. Clients are in my home network behind nat.
 In MySQL I have database asterisk with table sippeers, where I have
 clients added like this:
 INSERT INTO sippeers
 (name,defaultuser,host,sippasswd,fromuser,fromdomain,callbackextension,type)
 VALUES ('660', '660', 'dynamic', 'password', '660', 'testers.com
 ','660','friend');

 In this message there are some outputs and a sip trace of a register:
 https://www.mail-archive.com/sr-users@lists.sip-router.org/msg18558.html

 What I don't know is how to configure sip.conf, so far I've just been
 making guesses based on online examples and documentation.
 My current sip.conf looks like this:

 [general]
 bindport = 5070
 bindaddr = 127.0.0.1
 tcpbindaddr = 127.0.0.1:5070
 tcpenable = no
 limitonpeers = yes
 ;rtcachefriends = yes
 tos_sip=cs3
 tos_audio=ef
 realm = testers.com

 I've tried defining realm and domain values, but I lack proper
 understanding of those. Can you guys help me out? Are there any other
 configurations I need to check?

 

Re: [asterisk-users] Realtime integration: Unregistered clients showing as registered?

[Olli Heiskanen]

Hello,

After a small break from working on this, I got the idea of tcpdumping the
correct ports. What I see is REGISTER messages from Kamailio port to
Asterisk, which are replied with 401 Unauthorized. Why is this happening?
In my sippeers table the secret field has no value (tried both NULL and
empty string) and the added field sippasswd has the correct password for
the user.

The above might be the cause of my problem, would anyone be able to advice
me to get to correct behaviour? Now Kamailio sees the clients as
registered, which would be wrong if Asterisk doesn't.

cheers,
Olli



2014-04-24 11:27 GMT+03:00 Olli Heiskanen ohjelmistoarkkite...@gmail.com:


 Hello all,

 I've been testing a Kamailio Asterisk Realtime integration, and found a
 strange situation.

 My problem is that when using the integration, everything seems ok but
 Asterisk does not see the clients as registered. Kamailio and the clients
 report registered clients. Also calls fail.

 In Asterisk cli sip show peers shows nothing but for example realtime load
 sipusers name 660 shows the user data. Field regseconds has a value and
 fullcontact has value 'sip:660@127.0.0.1:5060' (kamailio ip:port as they
 are on the same machine).

 I have a very simple dialplan:

 [general]

 [default]
 exten = _XXX,1,NoOp(general : Dialed ${EXTEN})
  same = n,Dial(SIP/${EXTEN},3600,rt)
  same = n,Hangup


 Here's more on my problem and background to it, guys on the Kamailio list
 helped out but looks like I need to check my Asterisk configuration.
 https://www.mail-archive.com/sr-users@lists.sip-router.org/msg18555.html

 My goal is to have all clients in the asterisk database, asterisk (one at
 this point, several later) handling the calls and Kamailio as proxy. In
 Kamailio I have the WITH_MULTIDOMAIN directive on but I'm using only one
 domain 'testers.com'.

 I have Asterisk 11.8.1 and Kamailio 4.2.0-dev4 on CentOS 6.5, all are on
 the same rental virtual server. Clients are in my home network behind nat.
 In MySQL I have database asterisk with table sippeers, where I have
 clients added like this:
 INSERT INTO sippeers
 (name,defaultuser,host,sippasswd,fromuser,fromdomain,callbackextension,type)
 VALUES ('660', '660', 'dynamic', 'password', '660', 'testers.com
 ','660','friend');

 In this message there are some outputs and a sip trace of a register:
 https://www.mail-archive.com/sr-users@lists.sip-router.org/msg18558.html

 What I don't know is how to configure sip.conf, so far I've just been
 making guesses based on online examples and documentation.
 My current sip.conf looks like this:

 [general]
 bindport = 5070
 bindaddr = 127.0.0.1
 tcpbindaddr = 127.0.0.1:5070
 tcpenable = no
 limitonpeers = yes
 ;rtcachefriends = yes
 tos_sip=cs3
 tos_audio=ef
 realm = testers.com

 I've tried defining realm and domain values, but I lack proper
 understanding of those. Can you guys help me out? Are there any other
 configurations I need to check?

 Respectfully,
 Olli



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[asterisk-users] Realtime peers and sendrpid

[Ishfaq Malik]

Hello all

If I look at the sip peers table definition as provided with the source
of asterisk-1.8.23.0/ (looking at
contrib/realtime/mysql/sippeers.sql) for the sendrpid column it's an enum
with 2 possible values, yes and no.

However, the sip.conf allows 4 values, no, yes, rpid and pai.

Is this discrepancy an oversight? Is it possible to set the system default
to pai but an individual peer to rpid via a realtime table?

I have tried setting the system value to pai and a single peer value to yes
but it still sent pai rather than rpid.

Thanks in Advance

Ish
-- 

Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
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[asterisk-users] Realtime Pattern Matching

[Nick Olsen]

Hello All, Looking for a little guidance on Real Time Pattern Matching.
  
 We are attempting to block outbound 411 via when someone dials 
NXX-555-, The must common being NXX-555-1212. However, We have some 
outbound providers that consider any call to NXX-555- a directory 
assistance call. So simply making my pattern _NXX5551212 doesn't work.
  
 So as you can see from the lines below. If I Dial 321-555-1212 the call is 
being applied to _321NXX not my _NXX555. I assume because they are 
equally specific. Does anyone have any creative ideas to pattern match 
NXX-555- besides what I've done here?
  
 1056outbound_NXX5551Gotooutbound-411,411,1
Block Dir Assist
1057outbound_1NXX5551Gotooutbound-411,411,1
Block Dir Assist
1776outbound_321NXX1Gotooutbound-cocoa,${EXTEN},1   
 Outbound 321 Catchall
1777outbound_1321NXX1Gotooutbound-cocoa,${EXTEN},1  
  Outbound 1321 Catchall
  
  
 Thanks
  
 Nick Olsen
Network Operations  (855) FLSPEED  x106


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Re: [asterisk-users] Realtime Pattern Matching

[Richard Mudgett]

On Mon, May 12, 2014 at 12:25 PM, Nick Olsen n...@flhsi.com wrote:

 Hello All, Looking for a little guidance on Real Time Pattern Matching.

 We are attempting to block outbound 411 via when someone dials
 NXX-555-, The must common being NXX-555-1212. However, We have some
 outbound providers that consider any call to NXX-555- a directory
 assistance call. So simply making my pattern _NXX5551212 doesn't work.

 So as you can see from the lines below. If I Dial 321-555-1212 the call is
 being applied to _321NXX not my _NXX555. I assume because they are
 equally specific. Does anyone have any creative ideas to pattern match
 NXX-555- besides what I've done here?

 1056outbound_NXX5551Gotooutbound-411,411,1
  Block Dir Assist
 1057outbound_1NXX5551Gotooutbound-411,411,1
  Block Dir Assist
 1776outbound_321NXX1Goto
  outbound-cocoa,${EXTEN},1Outbound 321 Catchall
 1777outbound_1321NXX1Goto
  outbound-cocoa,${EXTEN},1Outbound 1321 Catchall


The _321NXX is more specific than _NXX555 that is why it is
selected.

See
https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching
for the rules of pattern matching.

Richard
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Re: [asterisk-users] Realtime Pattern Matching

[Josh Metzger]

On Mon, May 12, 2014 at 1:25 PM, Nick Olsen n...@flhsi.com wrote:

 Hello All, Looking for a little guidance on Real Time Pattern Matching.

 We are attempting to block outbound 411 via when someone dials
 NXX-555-, The must common being NXX-555-1212. However, We have some
 outbound providers that consider any call to NXX-555- a directory
 assistance call. So simply making my pattern _NXX5551212 doesn't work.

 So as you can see from the lines below. If I Dial 321-555-1212 the call
 is being applied to _321NXX not my _NXX555. I assume because they
 are equally specific. Does anyone have any creative ideas to pattern match
 NXX-555- besides what I've done here?

 1056outbound_NXX5551Gotooutbound-411,411,1
  Block Dir Assist
 1057outbound_1NXX5551Gotooutbound-411,411,1
  Block Dir Assist
 1776outbound_321NXX1Goto
  outbound-cocoa,${EXTEN},1Outbound 321 Catchall
 1777outbound_1321NXX1Goto
  outbound-cocoa,${EXTEN},1Outbound 1321 Catchall


If you're always trying to catch 555 for the prefix, instead of playing
with pattern matching the extension, you could test on the substring:

exten = _1NXXNXX,1,GotoIf($[${EXTEN:-7:3} =
555]?outbound-411,411,1)

This example would match any 1+Area code+number where the prefix is 555.
You could play with your pattern match to catch call to 1+AC+Number and
just AC+Number using this same test since it's right-delimited.

Josh
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Re: [asterisk-users] Realtime Pattern Matching

[Nick Olsen]

Thanks for the info everyone.
  
 With our particular dial plan. We break out _321 _305 _407.. Local area 
codes, And handle them differently. Where is _NXX555  works for numbers 
not specifically broken out. The fix for this was adding more specific 
_321555 for those handful of areacodes. Letting _NXX555 handle the 
rest.
  
 Thanks to everyone that replied.
  
 Nick Olsen
Network Operations  (855) FLSPEED  x106

  


 From: Josh Metzger joshdmetz...@gmail.com
Sent: Monday, May 12, 2014 1:43 PM
To: n...@flhsi.com, Asterisk Users Mailing List - Non-Commercial 
Discussion asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Realtime Pattern Matching   
   On Mon, May 12, 2014 at 1:25 PM, Nick Olsen n...@flhsi.com wrote:   
Hello All, Looking for a little guidance on Real Time Pattern Matching.
  
 We are attempting to block outbound 411 via when someone dials 
NXX-555-, The must common being NXX-555-1212. However, We have some 
outbound providers that consider any call to NXX-555- a directory 
assistance call. So simply making my pattern _NXX5551212 doesn't work.
  
 So as you can see from the lines below. If I Dial 321-555-1212 the call is 
being applied to _321NXX not my _NXX555. I assume because they are 
equally specific. Does anyone have any creative ideas to pattern match 
NXX-555- besides what I've done here?
  
 1056outbound_NXX5551Gotooutbound-411,411,1
Block Dir Assist
1057outbound_1NXX5551Gotooutbound-411,411,1
Block Dir Assist
1776outbound_321NXX1Gotooutbound-cocoa,${EXTEN},1   
 Outbound 321 Catchall
1777outbound_1321NXX1Gotooutbound-cocoa,${EXTEN},1  
  Outbound 1321 Catchall

 If you're always trying to catch 555 for the prefix, instead of playing 
with pattern matching the extension, you could test on the substring:
 
 exten = _1NXXNXX,1,GotoIf($[${EXTEN:-7:3} = 
555]?outbound-411,411,1)
 
 This example would match any 1+Area code+number where the prefix is 
555. You could play with your pattern match to catch call to 1+AC+Number 
and just AC+Number using this same test since it's right-delimited. 
 
 Josh


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[asterisk-users] Realtime integration: Unregistered clients showing as registered?

[Olli Heiskanen]

Hello all,

I've been testing a Kamailio Asterisk Realtime integration, and found a
strange situation.

My problem is that when using the integration, everything seems ok but
Asterisk does not see the clients as registered. Kamailio and the clients
report registered clients. Also calls fail.

In Asterisk cli sip show peers shows nothing but for example realtime load
sipusers name 660 shows the user data. Field regseconds has a value and
fullcontact has value 'sip:660@127.0.0.1:5060' (kamailio ip:port as they
are on the same machine).

I have a very simple dialplan:

[general]

[default]
exten = _XXX,1,NoOp(general : Dialed ${EXTEN})
 same = n,Dial(SIP/${EXTEN},3600,rt)
 same = n,Hangup


Here's more on my problem and background to it, guys on the Kamailio list
helped out but looks like I need to check my Asterisk configuration.
https://www.mail-archive.com/sr-users@lists.sip-router.org/msg18555.html

My goal is to have all clients in the asterisk database, asterisk (one at
this point, several later) handling the calls and Kamailio as proxy. In
Kamailio I have the WITH_MULTIDOMAIN directive on but I'm using only one
domain 'testers.com'.

I have Asterisk 11.8.1 and Kamailio 4.2.0-dev4 on CentOS 6.5, all are on
the same rental virtual server. Clients are in my home network behind nat.
In MySQL I have database asterisk with table sippeers, where I have clients
added like this:
INSERT INTO sippeers
(name,defaultuser,host,sippasswd,fromuser,fromdomain,callbackextension,type)
VALUES ('660', '660', 'dynamic', 'password', '660', 'testers.com
','660','friend');

In this message there are some outputs and a sip trace of a register:
https://www.mail-archive.com/sr-users@lists.sip-router.org/msg18558.html

What I don't know is how to configure sip.conf, so far I've just been
making guesses based on online examples and documentation.
My current sip.conf looks like this:

[general]
bindport = 5070
bindaddr = 127.0.0.1
tcpbindaddr = 127.0.0.1:5070
tcpenable = no
limitonpeers = yes
;rtcachefriends = yes
tos_sip=cs3
tos_audio=ef
realm = testers.com

I've tried defining realm and domain values, but I lack proper
understanding of those. Can you guys help me out? Are there any other
configurations I need to check?

Respectfully,
Olli
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[asterisk-users] Realtime in asterisk 12 removed/deprecated?

[Bruce Ferrell]

I was just told that realtime was no longer in asterisk 12, but I find 
enhancements in 12.2-rc2 and no sign in the wiki that this is true.

Can someone comment?


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Re: [asterisk-users] Realtime in asterisk 12 removed/deprecated?

[Eric Wieling]

All significant changes should be listed in the UPGRADE*.txt included in the 
Asterisk source code.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce Ferrell
Sent: Thursday, April 17, 2014 4:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Realtime in asterisk 12 removed/deprecated?

I was just told that realtime was no longer in asterisk 12, but I find 
enhancements in 12.2-rc2 and no sign in the wiki that this is true.

Can someone comment?


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Re: [asterisk-users] Realtime in asterisk 12 removed/deprecated?

[Bruce Ferrell]

Yeah, and I didn't find anything there.  I was looking for something a little more 
concrete that it should be...



On 04/17/2014 01:16 PM, Eric Wieling wrote:

All significant changes should be listed in the UPGRADE*.txt included in the 
Asterisk source code.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce Ferrell
Sent: Thursday, April 17, 2014 4:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Realtime in asterisk 12 removed/deprecated?

I was just told that realtime was no longer in asterisk 12, but I find 
enhancements in 12.2-rc2 and no sign in the wiki that this is true.

Can someone comment?


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Re: [asterisk-users] Realtime in asterisk 12 removed/deprecated?

[Joshua Colp]

Bruce Ferrell wrote:

I was just told that realtime was no longer in asterisk 12, but I find
enhancements in 12.2-rc2 and no sign in the wiki that this is true.

Can someone comment?


Realtime has not been removed or deprecated. A new model for newly 
written modules has been created, but nothing existing has been migrated 
to it or even will be (it's a fundamentally shift).


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

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Re: [asterisk-users] Realtime in asterisk 12 removed/deprecated?

[Bruce Ferrell]

On 04/17/2014 01:34 PM, Joshua Colp wrote:

Bruce Ferrell wrote:

I was just told that realtime was no longer in asterisk 12, but I find
enhancements in 12.2-rc2 and no sign in the wiki that this is true.

Can someone comment?


Realtime has not been removed or deprecated. A new model for newly written modules has been created, but nothing existing has been migrated to it or even will be (it's a 
fundamentally shift).



Thanks Joshua!

Always good to have a definitive statements

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Re: [asterisk-users] Realtime in asterisk 12 removed/deprecated?

[Joshua Colp]

Bruce Ferrell wrote:

On 04/17/2014 01:34 PM, Joshua Colp wrote:

Bruce Ferrell wrote:

I was just told that realtime was no longer in asterisk 12, but I find
enhancements in 12.2-rc2 and no sign in the wiki that this is true.

Can someone comment?


Realtime has not been removed or deprecated. A new model for newly
written modules has been created, but nothing existing has been
migrated to it or even will be (it's a fundamentally shift).


Thanks Joshua!

Always good to have a definitive statements


You're welcome. And fundamentally? What the heck was I thinking. 
*fundamental* shift.


Cheers,

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

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Re: [asterisk-users] Realtime Call Queues : call members in certain order

[Steven Wheeler]

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Thursday, February 27, 2014 7:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime Call Queues : call members in certain 
order

On 13-02-14 17:33, Steven Wheeler wrote:

From: 
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Thursday, February 13, 2014 7:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime Call Queues : call members in certain 
order


On 12-02-14 16:58, Steven Wheeler wrote:
From: 
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Wednesday, February 12, 2014 3:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Realtime Call Queues : call members in certain order

Hello,

I'm using MySQL realtime Call Queues (table queues and table queue_members).

I would like to ring the members of the call queue in a certain order. 
Therefore I use ring strategy lineair and I put the members into the table 
queue_members in the order in which they have to be rang.


So I have the queue :

| name   | musicclass | announce | context | timeout | monitor_type | 
monitor_format | queue_youarenext | queue_thereare | queue_callswaiting | 
queue_holdtime | queue_minutes | queue_seconds | queue_lessthan | 
queue_thankyou | queue_reporthold | announce_frequency | announce_round_seconds 
| announce_holdtime | announce_position | retry | wrapuptime | maxlen | 
servicelevel | strategy | joinempty | leavewhenempty | eventmemberstatus | 
eventwhencalled | reportholdtime | memberdelay | weight | timeoutrestart | 
periodic_announce | periodic_announce_frequency | ringinuse |
+++--+-+-+--++--++++---+---+++--+++---+---+---+++--+--+---++---+-++-+++---+-+---+
| queue6 | default| NULL | |  12 | NULL | NULL  
 | NULL | NULL   | NULL   | NULL   
| NULL  | NULL  | NULL   | NULL   | NULL
 | 30 |   NULL | No| yes
   | 5 | 10 |  0 | NULL | linear   | strict
| strict | NULL  | NULL|   NULL |   
 NULL |   NULL | no |   |   
0 | no|
+++--+-+-+--++--++++---+---+++--+++---+---+---+++--+--+---++---+-++-+++---+-+---+


and queue members :

+--++++-++
| uniqueid | membername | queue_name | interface  | penalty | 
paused |
+--++++-++
|   44 | queuemem4  | queue6 | SIP/queuemem4  |   0 |   NULL |
|   45 | queuemem2  | queue6 | SIP/queuemem2  |   0 |   NULL |
|   46 | queuemem5  | queue6 | SIP/queuemem5  |   0 |   NULL |
|   47 | queuemem1  | queue6 | SIP/queuemem1  |   0 |   NULL |
|   48 | queuemem10 | queue6 | SIP/queuemem10 |   0 |   NULL |
|   49 | queuemem18 | queue6 | SIP/queuemem18 |   0 |   NULL |
|   50 | queuemem17 | queue6 | SIP/queuemem17 |   0 |   NULL |
|   51 | queuemem12 | queue6 | SIP/queuemem12 |   0 |   NULL |
|   52 | queuemem16 | queue6 | SIP/queuemem16 |   0 |   NULL |
|   53 | queuemem13 | queue6 | SIP/queuemem13 |   0 |   NULL |
+--++++-++



You can see that the member queuemem4 is first in line to be rang (has the 
first and lowest uniqueid in the table).

But the first member that is being rang

Re: [asterisk-users] Realtime Call Queues : call members in certain order

[Jonas Kellens]

On 13-02-14 17:33, Steven Wheeler wrote:


*From:*asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas 
Kellens

*Sent:* Thursday, February 13, 2014 7:12 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Realtime Call Queues : call members in 
certain order


On 12-02-14 16:58, Steven Wheeler wrote:

*From:*asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
*Jonas Kellens
*Sent:* Wednesday, February 12, 2014 3:46 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Realtime Call Queues : call members in
certain order

Hello,

I'm using MySQL realtime Call Queues (table /queues/ and table
/queue_members/).

I would like to ring the members of the call queue in a certain
order. Therefore I use ring strategy /lineair /and I put the
members into the table /queue_members/ in the order in which they
have to be rang.


So I have the queue :

| name   | musicclass | announce | context | timeout |
monitor_type | monitor_format | queue_youarenext | queue_thereare
| queue_callswaiting | queue_holdtime | queue_minutes |
queue_seconds | queue_lessthan | queue_thankyou | queue_reporthold
| announce_frequency | announce_round_seconds | announce_holdtime
| announce_position | retry | wrapuptime | maxlen | servicelevel |
strategy | joinempty | leavewhenempty | eventmemberstatus |
eventwhencalled | reportholdtime | memberdelay | weight |
timeoutrestart | periodic_announce | periodic_announce_frequency |
ringinuse |

+++--+-+-+--++--++++---+---+++--+++---+---+---+++--+--+---++---+-++-+++---+-+---+
| queue6 | default| NULL | |  12 |
NULL | NULL   | NULL | NULL  
| NULL   | NULL   | NULL  |
NULL  | NULL   | NULL   | NULL
| 30 |   NULL | No   
| yes   | 5 | 10 |  0 | NULL |

linear   | strict| strict | NULL  |
NULL|   NULL |NULL |   NULL | no
|   |   0 | no|

+++--+-+-+--++--++++---+---+++--+++---+---+---+++--+--+---++---+-++-+++---+-+---+


and queue members :


+--++++-++
| uniqueid | membername | queue_name | interface 
| penalty | paused |


+--++++-++
|   44 | queuemem4  | queue6 | SIP/queuemem4  | 0 |   NULL |
|   45 | queuemem2  | queue6 | SIP/queuemem2  | 0 |   NULL |
|   46 | queuemem5  | queue6 | SIP/queuemem5  | 0 |   NULL |
|   47 | queuemem1  | queue6 | SIP/queuemem1  | 0 |   NULL |
|   48 | queuemem10 | queue6 | SIP/queuemem10 | 0 |   NULL |
|   49 | queuemem18 | queue6 | SIP/queuemem18 | 0 |   NULL |
|   50 | queuemem17 | queue6 | SIP/queuemem17 | 0 |   NULL |
|   51 | queuemem12 | queue6 | SIP/queuemem12 | 0 |   NULL |
|   52 | queuemem16 | queue6 | SIP/queuemem16 | 0 |   NULL |
|   53 | queuemem13 | queue6 | SIP/queuemem13 | 0 |   NULL |

+--++++-++



You can see that the member /queuemem4/ is first in line to be
rang (has the first and lowest uniqueid in the table).

But the first member that is being rang, is /queuemem1/. How come ??


Kind regards,

Jonas.

Jonas,

We encountered the same problem. It is a bug in the Queue
application. The Queue application actually orders

Re: [asterisk-users] Realtime Call Queues : call members in certain order

[Jonas Kellens]

On 12-02-14 16:58, Steven Wheeler wrote:


*From:*asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas 
Kellens

*Sent:* Wednesday, February 12, 2014 3:46 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Realtime Call Queues : call members in 
certain order


Hello,

I'm using MySQL realtime Call Queues (table /queues/ and table 
/queue_members/).


I would like to ring the members of the call queue in a certain order. 
Therefore I use ring strategy /lineair /and I put the members into the 
table /queue_members/ in the order in which they have to be rang.



So I have the queue :

| name   | musicclass | announce | context | timeout | 
monitor_type | monitor_format | queue_youarenext | queue_thereare | 
queue_callswaiting | queue_holdtime | queue_minutes | queue_seconds | 
queue_lessthan | queue_thankyou | queue_reporthold | 
announce_frequency | announce_round_seconds | announce_holdtime | 
announce_position | retry | wrapuptime | maxlen | servicelevel | 
strategy | joinempty | leavewhenempty | eventmemberstatus | 
eventwhencalled | reportholdtime | memberdelay | weight | 
timeoutrestart | periodic_announce | periodic_announce_frequency | 
ringinuse |

+++--+-+-+--++--++++---+---+++--+++---+---+---+++--+--+---++---+-++-+++---+-+---+
| queue6 | default| NULL | |  12 | NULL | 
NULL   | NULL | NULL   | 
NULL   | NULL   | NULL  | NULL  | 
NULL   | NULL   | NULL | 
30 |   NULL | No| yes   
| 5 | 10 |  0 | NULL | linear   | strict| 
strict | NULL  | NULL|   NULL 
|NULL |   NULL | no | |   
0 | no|

+++--+-+-+--++--++++---+---+++--+++---+---+---+++--+--+---++---+-++-+++---+-+---+


and queue members :

+--++++-++
| uniqueid | membername | queue_name | interface  | 
penalty | paused |

+--++++-++
|   44 | queuemem4  | queue6 | SIP/queuemem4  |   0 |   NULL |
|   45 | queuemem2  | queue6 | SIP/queuemem2  |   0 |   NULL |
|   46 | queuemem5  | queue6 | SIP/queuemem5  |   0 |   NULL |
|   47 | queuemem1  | queue6 | SIP/queuemem1  |   0 |   NULL |
|   48 | queuemem10 | queue6 | SIP/queuemem10 |   0 |   NULL |
|   49 | queuemem18 | queue6 | SIP/queuemem18 |   0 |   NULL |
|   50 | queuemem17 | queue6 | SIP/queuemem17 |   0 |   NULL |
|   51 | queuemem12 | queue6 | SIP/queuemem12 |   0 |   NULL |
|   52 | queuemem16 | queue6 | SIP/queuemem16 |   0 |   NULL |
|   53 | queuemem13 | queue6 | SIP/queuemem13 |   0 |   NULL |
+--++++-++



You can see that the member /queuemem4/ is first in line to be rang 
(has the first and lowest uniqueid in the table).


But the first member that is being rang, is /queuemem1/. How come ??


Kind regards,

Jonas.

Jonas,

We encountered the same problem. It is a bug in the Queue application. 
The Queue application actually orders members by their interface 
value. Here is the bug report I opened 
https://issues.asterisk.org/jira/browse/ASTERISK-18480 
https://issues.asterisk.org/jira/browse/ASTERISK-18480 which was 
closed as Not A Bug by Digium.  We worked around this by prepending 
an integer (001__, 002__, ...) to the interface in the database table 
and then removing it later in the dial plan. Hope this helps.


Steven Wheeler



Hello,

thank you for your reply.


Is it the membername or the interface that needs to be sorted to 
have a certain order in the call queue ?



How do you

Re: [asterisk-users] Realtime Call Queues : call members in certain order

[Steven Wheeler]

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Thursday, February 13, 2014 7:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime Call Queues : call members in certain 
order


On 12-02-14 16:58, Steven Wheeler wrote:
From: 
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Wednesday, February 12, 2014 3:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Realtime Call Queues : call members in certain order

Hello,

I'm using MySQL realtime Call Queues (table queues and table queue_members).

I would like to ring the members of the call queue in a certain order. 
Therefore I use ring strategy lineair and I put the members into the table 
queue_members in the order in which they have to be rang.


So I have the queue :

| name   | musicclass | announce | context | timeout | monitor_type | 
monitor_format | queue_youarenext | queue_thereare | queue_callswaiting | 
queue_holdtime | queue_minutes | queue_seconds | queue_lessthan | 
queue_thankyou | queue_reporthold | announce_frequency | announce_round_seconds 
| announce_holdtime | announce_position | retry | wrapuptime | maxlen | 
servicelevel | strategy | joinempty | leavewhenempty | eventmemberstatus | 
eventwhencalled | reportholdtime | memberdelay | weight | timeoutrestart | 
periodic_announce | periodic_announce_frequency | ringinuse |
+++--+-+-+--++--++++---+---+++--+++---+---+---+++--+--+---++---+-++-+++---+-+---+
| queue6 | default| NULL | |  12 | NULL | NULL  
 | NULL | NULL   | NULL   | NULL   
| NULL  | NULL  | NULL   | NULL   | NULL
 | 30 |   NULL | No| yes
   | 5 | 10 |  0 | NULL | linear   | strict
| strict | NULL  | NULL|   NULL |   
 NULL |   NULL | no |   |   
0 | no|
+++--+-+-+--++--++++---+---+++--+++---+---+---+++--+--+---++---+-++-+++---+-+---+


and queue members :

+--++++-++
| uniqueid | membername | queue_name | interface  | penalty | 
paused |
+--++++-++
|   44 | queuemem4  | queue6 | SIP/queuemem4  |   0 |   NULL |
|   45 | queuemem2  | queue6 | SIP/queuemem2  |   0 |   NULL |
|   46 | queuemem5  | queue6 | SIP/queuemem5  |   0 |   NULL |
|   47 | queuemem1  | queue6 | SIP/queuemem1  |   0 |   NULL |
|   48 | queuemem10 | queue6 | SIP/queuemem10 |   0 |   NULL |
|   49 | queuemem18 | queue6 | SIP/queuemem18 |   0 |   NULL |
|   50 | queuemem17 | queue6 | SIP/queuemem17 |   0 |   NULL |
|   51 | queuemem12 | queue6 | SIP/queuemem12 |   0 |   NULL |
|   52 | queuemem16 | queue6 | SIP/queuemem16 |   0 |   NULL |
|   53 | queuemem13 | queue6 | SIP/queuemem13 |   0 |   NULL |
+--++++-++



You can see that the member queuemem4 is first in line to be rang (has the 
first and lowest uniqueid in the table).

But the first member that is being rang, is queuemem1. How come ??


Kind regards,

Jonas.

Jonas,
We encountered the same problem. It is a bug in the Queue application. The 
Queue application actually orders members by their interface value. Here is the 
bug report I opened https://issues.asterisk.org/jira/browse/ASTERISK-18480 
which was closed as Not A Bug by Digium.  We worked around this by prepending 
an integer (001__, 002__

[asterisk-users] Realtime Call Queues : call members in certain order

[Jonas Kellens]

Hello,

I'm using MySQL realtime Call Queues (table /queues/ and table 
/queue_members/).


I would like to ring the members of the call queue in a certain order. 
Therefore I use ring strategy /lineair /and I put the members into the 
table /queue_members/ in the order in which they have to be rang.



So I have the queue :

| name   | musicclass | announce | context | timeout | 
monitor_type | monitor_format | queue_youarenext | queue_thereare | 
queue_callswaiting | queue_holdtime | queue_minutes | queue_seconds | 
queue_lessthan | queue_thankyou | queue_reporthold | announce_frequency 
| announce_round_seconds | announce_holdtime | announce_position | retry 
| wrapuptime | maxlen | servicelevel | strategy | joinempty | 
leavewhenempty | eventmemberstatus | eventwhencalled | reportholdtime | 
memberdelay | weight | timeoutrestart | periodic_announce | 
periodic_announce_frequency | ringinuse |

+++--+-+-+--++--++++---+---+++--+++---+---+---+++--+--+---++---+-++-+++---+-+---+
| queue6 | default| NULL | |  12 | NULL | 
NULL   | NULL | NULL | NULL   | 
NULL   | NULL  | NULL  | NULL   | 
NULL   | NULL | 30 |   NULL | No 
| yes   | 5 | 10 |  0 | NULL | 
linear   | strict| strict | NULL  | 
NULL|   NULL |NULL |   NULL | no 
|   |   0 | no|

+++--+-+-+--++--++++---+---+++--+++---+---+---+++--+--+---++---+-++-+++---+-+---+


and queue members :

+--++++-++
| uniqueid | membername | queue_name | interface | penalty | 
paused |

+--++++-++
|   44 | queuemem4  | queue6 | SIP/queuemem4  |   0 | NULL |
|   45 | queuemem2  | queue6 | SIP/queuemem2  |   0 | NULL |
|   46 | queuemem5  | queue6 | SIP/queuemem5  |   0 | NULL |
|   47 | queuemem1  | queue6 | SIP/queuemem1  |   0 | NULL |
|   48 | queuemem10 | queue6 | SIP/queuemem10 |   0 | NULL |
|   49 | queuemem18 | queue6 | SIP/queuemem18 |   0 | NULL |
|   50 | queuemem17 | queue6 | SIP/queuemem17 |   0 | NULL |
|   51 | queuemem12 | queue6 | SIP/queuemem12 |   0 | NULL |
|   52 | queuemem16 | queue6 | SIP/queuemem16 |   0 | NULL |
|   53 | queuemem13 | queue6 | SIP/queuemem13 |   0 | NULL |
+--++++-++



You can see that the member /queuemem4/ is first in line to be rang (has 
the first and lowest uniqueid in the table).


But the first member that is being rang, is /queuemem//1/. How come ??


Kind regards,

Jonas.

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Re: [asterisk-users] Realtime Call Queues : call members in certain order

[Steven Wheeler]

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Wednesday, February 12, 2014 3:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Realtime Call Queues : call members in certain order

Hello,

I'm using MySQL realtime Call Queues (table queues and table queue_members).

I would like to ring the members of the call queue in a certain order. 
Therefore I use ring strategy lineair and I put the members into the table 
queue_members in the order in which they have to be rang.


So I have the queue :

| name   | musicclass | announce | context | timeout | monitor_type | 
monitor_format | queue_youarenext | queue_thereare | queue_callswaiting | 
queue_holdtime | queue_minutes | queue_seconds | queue_lessthan | 
queue_thankyou | queue_reporthold | announce_frequency | announce_round_seconds 
| announce_holdtime | announce_position | retry | wrapuptime | maxlen | 
servicelevel | strategy | joinempty | leavewhenempty | eventmemberstatus | 
eventwhencalled | reportholdtime | memberdelay | weight | timeoutrestart | 
periodic_announce | periodic_announce_frequency | ringinuse |
+++--+-+-+--++--++++---+---+++--+++---+---+---+++--+--+---++---+-++-+++---+-+---+
| queue6 | default| NULL | |  12 | NULL | NULL  
 | NULL | NULL   | NULL   | NULL   
| NULL  | NULL  | NULL   | NULL   | NULL
 | 30 |   NULL | No| yes
   | 5 | 10 |  0 | NULL | linear   | strict
| strict | NULL  | NULL|   NULL |   
 NULL |   NULL | no |   |   
0 | no|
+++--+-+-+--++--++++---+---+++--+++---+---+---+++--+--+---++---+-++-+++---+-+---+


and queue members :

+--++++-++
| uniqueid | membername | queue_name | interface  | penalty | 
paused |
+--++++-++
|   44 | queuemem4  | queue6 | SIP/queuemem4  |   0 |   NULL |
|   45 | queuemem2  | queue6 | SIP/queuemem2  |   0 |   NULL |
|   46 | queuemem5  | queue6 | SIP/queuemem5  |   0 |   NULL |
|   47 | queuemem1  | queue6 | SIP/queuemem1  |   0 |   NULL |
|   48 | queuemem10 | queue6 | SIP/queuemem10 |   0 |   NULL |
|   49 | queuemem18 | queue6 | SIP/queuemem18 |   0 |   NULL |
|   50 | queuemem17 | queue6 | SIP/queuemem17 |   0 |   NULL |
|   51 | queuemem12 | queue6 | SIP/queuemem12 |   0 |   NULL |
|   52 | queuemem16 | queue6 | SIP/queuemem16 |   0 |   NULL |
|   53 | queuemem13 | queue6 | SIP/queuemem13 |   0 |   NULL |
+--++++-++



You can see that the member queuemem4 is first in line to be rang (has the 
first and lowest uniqueid in the table).

But the first member that is being rang, is queuemem1. How come ??


Kind regards,

Jonas.

Jonas,
We encountered the same problem. It is a bug in the Queue application. The 
Queue application actually orders members by their interface value. Here is the 
bug report I opened https://issues.asterisk.org/jira/browse/ASTERISK-18480 
which was closed as Not A Bug by Digium.  We worked around this by prepending 
an integer (001__, 002__, ...) to the interface in the database table and then 
removing it later in the dial plan. Hope this helps.
Steven Wheeler
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Re: [asterisk-users] Realtime Call Files

[Carlos Chavez]

On 10/31/13, 8:44 AM, Rizwan Hisham wrote:

Hi all,
Is there any way of originating calls in future without using call files?

We have 2 servers (1 active at a time). If we use call files with 
modification date in future, on the 1st server and it dies and, we 
activate the second server but we lose the call files.


I could have a cronjob on both servers and create callfiles reading 
execution time from database, but this involves some other complications.


Any crazy ideas would be helpful.

Thanks


The easiest way to do this would be with AMI and originate your 
calls in realtime.  That way you do not have to worry about which server 
will handle the call, the one you connect to will.


--
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Carlos Chávez
+52 (55)9116-91161


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[asterisk-users] Realtime Call Files

[Rizwan Hisham]

Hi all,
Is there any way of originating calls in future without using call files?

We have 2 servers (1 active at a time). If we use call files with
modification date in future, on the 1st server and it dies and, we activate
the second server but we lose the call files.

I could have a cronjob on both servers and create callfiles reading
execution time from database, but this involves some other complications.

Any crazy ideas would be helpful.

Thanks

-- 
Best Ragards
Rizwan H Qureshi

V: +971 (0) 528272154
linkedin.com/in/rhqureshi
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Re: [asterisk-users] Realtime Call Files

[jg]

There are several options: AMI and ARI (vs. 12). Depending on what you are trying to do there is 
also AGI and FastAGI.


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Re: [asterisk-users] Realtime Call Files

[Ioan Indreias]

Hi Rizwan ,

Have you tried to define astspooldir (usually /var/spool/asterisk) to a
shared filesystem? Or to create a symlink for outgoing directory (where the
call files have to be placed) to a directory placed on a shared filesystem
(eg on a NAS)?

Just brainstorming - not yet tried and maybe completely wrong :)

HTH,
Ioan


On Thu, Oct 31, 2013 at 4:44 PM, Rizwan Hisham rizwanhas...@gmail.comwrote:

 Hi all,
 Is there any way of originating calls in future without using call files?

 We have 2 servers (1 active at a time). If we use call files with
 modification date in future, on the 1st server and it dies and, we activate
 the second server but we lose the call files.

 I could have a cronjob on both servers and create callfiles reading
 execution time from database, but this involves some other complications.

 Any crazy ideas would be helpful.

 Thanks

 --
 Best Ragards
 Rizwan H Qureshi

 V: +971 (0) 528272154
 linkedin.com/in/rhqureshi



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[asterisk-users] realtime voicemail asterisk 11

[troxlinux]

Hi list, I'm trying to put my voicemail on asterisk realtime with 11.XX,
generate tables in a couple of files in the folder realtime / mysql ,
voicemail_messages.sql and voicemail.sql

the connection with mysql and odbc works well

isql asterisk useradmin xxx
+---+
| Connected!|
|   |
| sql-statement |
| help [tablename]  |
| quit  |
|   |
+---+



[Oct 14 10:06:16] WARNING[10037][C-0003]: res_odbc.c:645
ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42S02:
[MySQL][ODBC 5.2(w) Driver][mysqld-5.6.12]Table 'asterisk.voicemessages'
doesn't exist (86)
[Oct 14 10:06:16] WARNING[10037][C-0003]: res_odbc.c:657
ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying connection to
asterisk [asterisk]...
[Oct 14 10:06:16] WARNING[10037][C-0003]: res_odbc.c:761
ast_odbc_sanity_check: Connection is down attempting to reconnect...
[Oct 14 10:06:16] NOTICE[10037][C-0003]: res_odbc.c:1527
odbc_obj_connect: Connecting asterisk
[Oct 14 10:06:16] NOTICE[10037][C-0003]: res_odbc.c:1559
odbc_obj_connect: res_odbc: Connected to asterisk [asterisk]
[Oct 14 10:06:16] WARNING[10037][C-0003]: app_voicemail.c:5609
messagecount: SQL Execute error!


any help is appreciated!



-- 
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Re: [asterisk-users] realtime voicemail asterisk 11

[Warren Selby]

On Mon, Oct 14, 2013 at 11:13 AM, troxlinux xserverli...@gmail.com wrote:

 Hi list, I'm trying to put my voicemail on asterisk realtime with 11.XX,
 generate tables in a couple of files in the folder realtime / mysql ,
 voicemail_messages.sql and voicemail.sql

 the connection with mysql and odbc works well

 isql asterisk useradmin xxx
 +---+
 | Connected!|
 |   |
 | sql-statement |
 | help [tablename]  |
 | quit  |
 |   |
 +---+



 [Oct 14 10:06:16] WARNING[10037][C-0003]: res_odbc.c:645
 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42S02:
 [MySQL][ODBC 5.2(w) Driver][mysqld-5.6.12]Table 'asterisk.voicemessages'
 doesn't exist (86)
 [Oct 14 10:06:16] WARNING[10037][C-0003]: res_odbc.c:657
 ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying connection to
 asterisk [asterisk]...
 [Oct 14 10:06:16] WARNING[10037][C-0003]: res_odbc.c:761
 ast_odbc_sanity_check: Connection is down attempting to reconnect...
 [Oct 14 10:06:16] NOTICE[10037][C-0003]: res_odbc.c:1527
 odbc_obj_connect: Connecting asterisk
 [Oct 14 10:06:16] NOTICE[10037][C-0003]: res_odbc.c:1559
 odbc_obj_connect: res_odbc: Connected to asterisk [asterisk]
 [Oct 14 10:06:16] WARNING[10037][C-0003]: app_voicemail.c:5609
 messagecount: SQL Execute error!



Could you post a sanitized version of your res_config_mysql.conf and
extconfig.conf files?  I'm thinking maybe you've got an error in there
somewhere that's causing this error.


--
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Warren Selby, dCAP
http://www.SelbyTech.com http://www.selbytech.com
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Re: [asterisk-users] realtime voicemail asterisk 11

[troxlinux]

res_config_mysql

[general]
dbhost = 127.0.0.1
dbname = asterisk
dbuser = root
dbpass = x
dbport = 3306
dbsock = /var/lib/mysql/mysql.sock

extconfig.conf
voicemail= mysql,asterisk,voicemail_messages


right now I'm around trying to rename the table voicemail_messages by
asterisk.voicemessages


2013/10/14 Warren Selby wcse...@selbytech.com

 On Mon, Oct 14, 2013 at 11:13 AM, troxlinux xserverli...@gmail.comwrote:

 Hi list, I'm trying to put my voicemail on asterisk realtime with 11.XX,
 generate tables in a couple of files in the folder realtime / mysql ,
 voicemail_messages.sql and voicemail.sql

 the connection with mysql and odbc works well

 isql asterisk useradmin xxx
 +---+
 | Connected!|
 |   |
 | sql-statement |
 | help [tablename]  |
 | quit  |
 |   |
 +---+



 [Oct 14 10:06:16] WARNING[10037][C-0003]: res_odbc.c:645
 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42S02:
 [MySQL][ODBC 5.2(w) Driver][mysqld-5.6.12]Table 'asterisk.voicemessages'
 doesn't exist (86)
 [Oct 14 10:06:16] WARNING[10037][C-0003]: res_odbc.c:657
 ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying connection to
 asterisk [asterisk]...
 [Oct 14 10:06:16] WARNING[10037][C-0003]: res_odbc.c:761
 ast_odbc_sanity_check: Connection is down attempting to reconnect...
 [Oct 14 10:06:16] NOTICE[10037][C-0003]: res_odbc.c:1527
 odbc_obj_connect: Connecting asterisk
 [Oct 14 10:06:16] NOTICE[10037][C-0003]: res_odbc.c:1559
 odbc_obj_connect: res_odbc: Connected to asterisk [asterisk]
 [Oct 14 10:06:16] WARNING[10037][C-0003]: app_voicemail.c:5609
 messagecount: SQL Execute error!



 Could you post a sanitized version of your res_config_mysql.conf and
 extconfig.conf files?  I'm thinking maybe you've got an error in there
 somewhere that's causing this error.


 --
 Thanks,
 Warren Selby, dCAP
 http://www.SelbyTech.com http://www.selbytech.com


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Re: [asterisk-users] realtime voicemail asterisk 11

[Warren Selby]

On Mon, Oct 14, 2013 at 12:19 PM, troxlinux xserverli...@gmail.com wrote:

 res_config_mysql

 [general]
 dbhost = 127.0.0.1
 dbname = asterisk
 dbuser = root
 dbpass = x
 dbport = 3306
 dbsock = /var/lib/mysql/mysql.sock

 extconfig.conf
 voicemail= mysql,asterisk,voicemail_messages



First issue I see - you've got the context named [general] in
res_config_mysql, but you're attempting to connect to asterisk in
extconfig.conf (mysql,*asterisk*,voicemail_messages).  The second item in
extconfig.conf should match the database context name in res_config_mysql.
So either fix res_config_mysql by changing [general] to [asterisk], or fix
extconfig.conf by changing the line to voicemail =
mysql,general,voicemail_messages.

Make whichever change you prefer (I would make the change in
res_config_mysql.conf personally, but it's up to you), and then reload
asterisk to see if that resolves the error.  Otherwise, post whatever
you're new error is.


--
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Warren Selby, dCAP
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Re: [asterisk-users] realtime voicemail asterisk 11

[troxlinux]

thnk Warren , I only see one warning message

[Oct 14 13:12:14] WARNING[2736][C-]: app_voicemail.c:3768
retrieve_file: SQL Get Data error! coltitle=category
[SELECT * FROM voicemessages WHERE dir=? AND msgnum=?]





2013/10/14 Warren Selby wcse...@selbytech.com

 On Mon, Oct 14, 2013 at 12:19 PM, troxlinux xserverli...@gmail.comwrote:

 res_config_mysql

 [general]
 dbhost = 127.0.0.1
 dbname = asterisk
 dbuser = root
 dbpass = x
 dbport = 3306
 dbsock = /var/lib/mysql/mysql.sock

 extconfig.conf
 voicemail= mysql,asterisk,voicemail_messages



 First issue I see - you've got the context named [general] in
 res_config_mysql, but you're attempting to connect to asterisk in
 extconfig.conf (mysql,*asterisk*,voicemail_messages).  The second item in
 extconfig.conf should match the database context name in res_config_mysql.
 So either fix res_config_mysql by changing [general] to [asterisk], or fix
 extconfig.conf by changing the line to voicemail =
 mysql,general,voicemail_messages.

 Make whichever change you prefer (I would make the change in
 res_config_mysql.conf personally, but it's up to you), and then reload
 asterisk to see if that resolves the error.  Otherwise, post whatever
 you're new error is.


 --
 Thanks,
 Warren Selby, dCAP
 http://www.SelbyTech.com http://www.selbytech.com

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Re: [asterisk-users] realtime voicemail asterisk 11

[Warren Selby]

On Mon, Oct 14, 2013 at 2:15 PM, troxlinux xserverli...@gmail.com wrote:

 thnk Warren , I only see one warning message

 [Oct 14 13:12:14] WARNING[2736][C-]: app_voicemail.c:3768
 retrieve_file: SQL Get Data error! coltitle=category
 [SELECT * FROM voicemessages WHERE dir=? AND msgnum=?]


I'm not sure on this.  Hopefully someone else can help.


--
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http://www.SelbyTech.com http://www.selbytech.com
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Re: [asterisk-users] realtime voicemail asterisk 11

[troxlinux]

ok , thnk Warren , your help has been invaluable.


2013/10/14 Warren Selby wcse...@selbytech.com

 On Mon, Oct 14, 2013 at 2:15 PM, troxlinux xserverli...@gmail.com wrote:

 thnk Warren , I only see one warning message

 [Oct 14 13:12:14] WARNING[2736][C-]: app_voicemail.c:3768
 retrieve_file: SQL Get Data error! coltitle=category
 [SELECT * FROM voicemessages WHERE dir=? AND msgnum=?]


 I'm not sure on this.  Hopefully someone else can help.


 --
 Thanks,
 Warren Selby, dCAP
 http://www.SelbyTech.com http://www.selbytech.com


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[asterisk-users] Realtime Mysql

[Phibee Network Operation Center]

Hello,

I am looking to know if it is possible to modify the SQL query that is 
on Realtime sip accounts.


I want multiple servers use the same sql table, so getting an extra 
server field to indicate that the data is valid on the X server


is this possible?

thank you in advance
jerome

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Re: [asterisk-users] Realtime Mysql

[Gareth Blades]

On 27/09/13 17:47, Phibee Network Operation Center wrote:

Hello,

I am looking to know if it is possible to modify the SQL query that is 
on Realtime sip accounts.


I want multiple servers use the same sql table, so getting an extra 
server field to indicate that the data is valid on the X server


is this possible?

thank you in advance
jerome

I dont know but you should be able to at least create multiple mysql 
views with each one being referenced by the appropiate server.


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Re: [asterisk-users] realtime sip.conf and templates

[Olle E. Johansson]

6 jun 2013 kl. 17:41 skrev Daniel Pocock dan...@pocock.com.au:

 On 06/06/13 15:51, Daniel Pocock wrote:
 Is the template capability in sip.conf compatible with realtime sip.conf
 entries such as users in a database?
 
 I notice that contrib/realtime/mysql/sippeers.sql and the wiki page
 don't mention a template column:
 
 https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
 
 while some third-party examples do suggest that a column named
 template is permitted:
 
 http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip
 
 I have actually tried adding that column template into sippeers and
 setting the value as the name of a template from my sip.conf - on
 Asterisk 11.4, it seems to ignore the column.  If there is a way to do
 this, it would be useful to have it in the wiki.
 
The templates are part of the configuration file (text files) parser and 
not supported in databases.

/O

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[asterisk-users] realtime sip.conf and templates

[Daniel Pocock]

Is the template capability in sip.conf compatible with realtime sip.conf
entries such as users in a database?

I notice that contrib/realtime/mysql/sippeers.sql and the wiki page
don't mention a template column:

https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure

while some third-party examples do suggest that a column named
template is permitted:

http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip



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Re: [asterisk-users] realtime sip.conf and templates

[Daniel Pocock]

On 06/06/13 15:51, Daniel Pocock wrote:
 Is the template capability in sip.conf compatible with realtime sip.conf
 entries such as users in a database?

 I notice that contrib/realtime/mysql/sippeers.sql and the wiki page
 don't mention a template column:

 https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure

 while some third-party examples do suggest that a column named
 template is permitted:

 http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip

I have actually tried adding that column template into sippeers and
setting the value as the name of a template from my sip.conf - on
Asterisk 11.4, it seems to ignore the column.  If there is a way to do
this, it would be useful to have it in the wiki.





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[asterisk-users] realtime peer w/ callbackextension does not register after 'sip reload'

[Marie Fischer]

Hello everybody,

I am having a problem with realtime SIP peers.
On Asterisk 1.8, I had SIP peers for external SIP providers configured in 
database and additional register lines in sip.conf so they would register.
Now I upgraded to Asterisk 11.3.0, partly because of the promised 
callbackextension feature for realtime peers 
(https://reviewboard.asterisk.org/r/1717/). Removed the 'register' lines from 
sip.conf. My peers register correctly when Asterisk is started or if I do 
'module unload chan_sip.so; module load chan_sip.so', but if I do 'sip reload', 
they stay in 'Unregistered' state forever.

*CLI sip show registry
Hostdnsmgr Username   Refresh State 
   Reg.Time 
xxx.xxx.xxx.xxx:5060  N    45 
Registered   Fri, 05 Apr 2013 05:37:02
1 SIP registrations.
*CLI 
*CLI sip reload
*CLI  Reloading SIP
  == Parsing '/etc/asterisk/sip.conf': Found
  == Using SIP CoS mark 4
[Apr  5 05:37:59] NOTICE[16991]: chan_sip.c:5527 
register_realtime_peers_with_callbackextens: Created realtime peer 'peer' for 
registration
  == Parsing '/etc/asterisk/sip_notify.conf': Found
*CLI 
*CLI 
*CLI sip show registry
Hostdnsmgr Username   Refresh State 
   Reg.Time 
xxx.xxx.xxx.xxx:5060  N    60 
Unregistered  
1 SIP registrations.
*CLI 

Also, sip show peers shows the peer correctly after restart, but is empty 
after 'sip reload'.

If I add the register line back to sip.conf, I get 2 lines for the same peer 
(in 'sip show registry') and both show state = registered. Strange.

Tried to dig through the code in chan_sip.c and one difference seems to be in 
the register line created by build_peer() - it includes the peername 
(register = peer?user:secret@host/extension), whereas in my config file I had 
just register = user:secret@host/extension. I removed the peer part from the 
source and recompiled, and if I recall correctly the registration survived 
sip reload after that, but that's a hack, not a solution. :)

I found this bug: https://issues.asterisk.org/jira/browse/ASTERISK-20611, but I 
don't think that's my issue - anyway, it should be fixed by now, but I still 
had the same issue with 11.4.0-rc1.

Does anybody have experience with realtime peers registering using 
callbackextension? Does this problem seem like a configuration issue or should 
I file a bug report? 

Sorry if you read this twice, I am crossposting to http://forums.asterisk.org 
as I still haven't figured out the best place to get answers. ;)

-- 

marie




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Re: [asterisk-users] Realtime vs Static Files

[Dan Journo]

 It is really unbelievable ... I was thinking: Asterisk uses an internal 
 database to maintain states of peers. It is usually located in 
 /var/lib/asterisk/astdb and it is a berkely db, but other database backends 
 seem available. Are you sharing also this database between the two servers? 
 It is the only option left...

The only thing shared is the sip realtime db.

I think i'm going to try removing the sip realtime db and automate the creation 
of the sip.conf file and issuing of the 'sip reload' and see if the problem 
goes away.


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Re: [asterisk-users] Realtime vs Static Files

[Leandro Dardini]

It is a shame we were unable to find the solution to your problem. Do you
want to setup a test system like the good one and let me access it to check
what is going on? I am really really curious.

Leandro
Il giorno 26/gen/2013 19:49, Dan Journo d...@keshercommunications.com ha
scritto:

  It is really unbelievable ... I was thinking: Asterisk uses an internal
 database to maintain states of peers. It is usually located in
 /var/lib/asterisk/astdb and it is a berkely db, but other database backends
 seem available. Are you sharing also this database between the two servers?
 It is the only option left...

 ** **

 The only thing shared is the sip realtime db.

 ** **

 I think i'm going to try removing the sip realtime db and automate the
 creation of the sip.conf file and issuing of the 'sip reload' and see if
 the problem goes away.

 ** **

 ** **

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Re: [asterisk-users] Realtime vs Static Files

[Dan Journo]

 Upgrading to the latest version didn't help. After about 30 minutes, 
 Asterisk2 tries to send out OPTIONS keepalive packets to peers listed as 
 Registered on Asterisk1.
 It is something really amazing... Can you run sip show peers on each one of 
 the servers and post the response?

 You said the second asterisk is completely opaque to your peers. Can you run 
 a tcpdump on secondary server to see if for some obscure reason the phones 
 try to contact the secondary asterisk?

I'll monitor one peer using tcpdump over a few hours and then review the 
packets. However, SIP DEBUG isn't showing any REGISTER packets.

Here's the sip peers output. Values and names have been hidden. Some appear as 
Unreachable the secondary server and some appear as OK.
I think some are listed as OK because the endpoint routers are performing some 
type of SIP ALG and routing packets based on port number and not source ip 
address.
However, from the SIP DEBUG output, it seems clear that the secondary server in 
this example is sending out Keepalives based on the information that the 
primary server has entered into the realtime DB.

Show peers Output from a primary server
Name/username  HostDyn 
Forcerport ACL Port Status Realtime
a201/A201  217.x.x.48D   N 
65229OK (88 ms) Cached RT
a202 (Unspecified)D   N 
0UNREACHABLE Cached RT
b201/44845287  78.x.x.101   D   N 5060 
OK (26 ms) Cached RT
c201/s   193.x.x.174  D   N 5060
 OK (52 ms) Cached RT
d201/d201  94.x.x.228 D   N 5060 OK 
(33 ms) Cached RT
e201/e20194.x.x.44 D   N 55018  
  OK (40 ms) Cached RT
e202/e20294.x.x.44 D   N 55022  
  OK (46 ms) Cached RT
e203/e20394.x.x.44 D   N 55024  
  OK (40 ms) Cached RT
e204/e20494.x.x.44 D   N 55008  
  OK (40 ms) Cached RT
e205/e20594.x.x.44 D   N 55016  
  OK (41 ms) Cached RT
e206/e20694.x.x.44 D   N 55014  
  OK (40 ms) Cached RT
e207/e20794.x.x.44 D   N 55020  
  OK (41 ms) Cached RT
e208/e20894.x.x.44 D   N 5060   
  OK (41 ms) Cached RT
e209/e20994.x.x.44 D   N 55012  
  OK (40 ms) Cached RT
e210/e21094.x.x.44 D   N 55010  
  OK (41 ms) Cached RT
e211/e21194.x.x.44 D   N 55026  
  OK (38 ms) Cached RT
e212/e21281.x.x.93D   N 5060
 OK (46 ms) Cached RT
f201 (Unspecified)D   N 0   
 UNREACHABLE Cached RT
g201/g  78.x.x.207 D   N 5060 OK 
(29 ms) Cached RT
h201/h201  217.x.x.78   D   N 38980OK 
(22 ms) Cached RT
i201 (Unspecified)D   N 0   
 UNREACHABLE Cached RT
i203/ i203 109.x.x.103  D   N 5060 OK 
(32 ms) Cached RT
i204/ i204 109.x.x.103  D   N 1025 OK 
(31 ms) Cached RT
i205/ i205 81.x.x.144 D   N 5060 OK 
(32 ms) Cached RT
i206/ i206  109.x.x.103  D   N 1035 OK 
(31 ms) Cached RT
i207/ i207  109.x.x.103  D   N 1032 OK 
(32 ms) Cached RT
i208/ i208 109.x.x.103  D   N 1024 OK 
(31 ms) Cached RT
j201/s   94.x.x.62D   N 57813   
 OK (35 ms) Cached RT
o201/o201  92.x.x.86 D   N 51824
OK (47 ms) Cached RT
o202/o202  92.x.x.86 D   N 58641
OK (48 ms) Cached RT
o203/o203  92.x.x.86 D   N 49172
OK (47 ms) Cached RT
j204/j204  176.x.x.214  D   N 34824
OK (49 ms) Cached RT
k201/k201  2.x.x.169 D   N 52757OK 
(53 ms) Cached RT
k202/k202  (Unspecified)D   N 0
UNKNOWNCached RT
l201/l201(Unspecified)D   N 0   
 UNKNOWNCached RT
m201/s  92.x.x.95  D   N 54020  
  OK (32 ms) Cached RT
n201   (Unspecified)   

Re: [asterisk-users] Realtime vs Static Files

[Leandro Dardini]

2013/1/25 Dan Journo d...@keshercommunications.com

  Upgrading to the latest version didn't help. After about 30 minutes,
 Asterisk2 tries to send out OPTIONS keepalive packets to peers listed as
 Registered on Asterisk1.

  It is something really amazing... Can you run sip show peers on each
 one of the servers and post the response?

 ** **

  You said the second asterisk is completely opaque to your peers. Can
 you run a tcpdump on secondary server to see if for some obscure reason the
 phones try to contact the secondary asterisk?

 ** **

 I'll monitor one peer using tcpdump over a few hours and then review the
 packets. However, SIP DEBUG isn't showing any REGISTER packets.

 ** **

 Here's the sip peers output. Values and names have been hidden. Some
 appear as Unreachable the secondary server and some appear as OK.

 I think some are listed as OK because the endpoint routers are performing
 some type of SIP ALG and routing packets based on port number and not
 source ip address.

 However, from the SIP DEBUG output, it seems clear that the secondary
 server in this example is sending out Keepalives based on the information
 that the primary server has entered into the realtime DB.

 ** **

 *Show peers Output from a primary server*

 Name/username  HostDyn
 Forcerport ACL Port Status Realtime

 a201/A201  217.x.x.48D
 N 65229OK (88 ms) Cached RT

 a202 (Unspecified)D
 N 0UNREACHABLE Cached RT

 b201/44845287  78.x.x.101   D   N
 5060 OK (26 ms) Cached RT

 c201/s   193.x.x.174  D   N
 5060 OK (52 ms) Cached RT

 d201/d201  94.x.x.228 D   N
 5060 OK (33 ms) Cached RT

 e201/e20194.x.x.44 D   N
 55018OK (40 ms) Cached RT

 e202/e20294.x.x.44 D   N
 55022OK (46 ms) Cached RT

 e203/e20394.x.x.44 D   N
 55024OK (40 ms) Cached RT

 e204/e20494.x.x.44 D   N
 55008OK (40 ms) Cached RT

 e205/e20594.x.x.44 D   N
 55016OK (41 ms) Cached RT

 e206/e20694.x.x.44 D   N
 55014OK (40 ms) Cached RT

 e207/e20794.x.x.44 D   N
 55020OK (41 ms) Cached RT

 e208/e20894.x.x.44 D   N
 5060 OK (41 ms) Cached RT

 e209/e20994.x.x.44 D   N
 55012OK (40 ms) Cached RT

 e210/e21094.x.x.44 D   N
 55010OK (41 ms) Cached RT

 e211/e21194.x.x.44 D   N
 55026OK (38 ms) Cached RT

 e212/e21281.x.x.93D   N
 5060 OK (46 ms) Cached RT

 f201 (Unspecified)D   N
 0UNREACHABLE Cached RT

 g201/g  78.x.x.207 D   N 5060
 OK (29 ms) Cached RT

 h201/h201  217.x.x.78   D   N 38980
 OK (22 ms) Cached RT

 i201 (Unspecified)D   N
 0UNREACHABLE Cached RT

 i203/ i203 109.x.x.103  D   N 5060
 OK (32 ms) Cached RT

 i204/ i204 109.x.x.103  D   N 1025
 OK (31 ms) Cached RT

 i205/ i205 81.x.x.144 D   N
 5060 OK (32 ms) Cached RT

 i206/ i206  109.x.x.103  D   N 1035
  OK (31 ms) Cached RT

 i207/ i207  109.x.x.103  D   N
 1032 OK (32 ms) Cached RT

 i208/ i208 109.x.x.103  D   N 1024
 OK (31 ms) Cached RT

 j201/s   94.x.x.62D   N
 57813OK (35 ms) Cached RT

 o201/o201  92.x.x.86 D   N
 51824OK (47 ms) Cached RT

 o202/o202  92.x.x.86 D   N
 58641OK (48 ms) Cached RT

 o203/o203  92.x.x.86 D   N
 49172OK (47 ms) Cached RT

 j204/j204  176.x.x.214  D   N
 34824OK (49 ms) Cached RT

 k201/k201  2.x.x.169 D   N
 52757OK (53 ms) Cached RT

 k202/k202  (Unspecified)D   N
 0UNKNOWNCached RT

 l201/l201(Unspecified)D   N
 0UNKNOWNCached RT

 m201/s  92.x.x.95  D   N
 54020OK (32 ms) Cached RT

 n201   (Unspecified) 

Re: [asterisk-users] Realtime vs Static Files

[Dan Journo]

 I am curious, is your version of asterisk correctly compiling the regserver 
 field? Each server needs to have a distinct server name.

Upgrading to the latest version didn't help. After about 30 minutes, Asterisk2 
tries to send out OPTIONS keepalive packets to peers listed as Registered on 
Asterisk1.
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Re: [asterisk-users] Realtime vs Static Files

[Leandro Dardini]

2013/1/24 Dan Journo d...@keshercommunications.com

  I am curious, is your version of asterisk correctly compiling the
 regserver field? Each server needs to have a distinct server name.

 ** **

 Upgrading to the latest version didn't help. After about 30 minutes,
 Asterisk2 tries to send out OPTIONS keepalive packets to peers listed as
 Registered on Asterisk1.


It is something really amazing... Can you run sip show peers on each one
of the servers and post the response?

You said the second asterisk is completely opaque to your peers. Can you
run a tcpdump on secondary server to see if for some obscure reason the
phones try to contact the secondary asterisk?

Leandro
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[asterisk-users] Realtime vs Static Files

[Dan Journo]

Hi,

We're trying to decide whether to switch back to a static file for sip.conf. 
Currently we use mysql realtime but can't see any real benefit.

Why would someone choose realtime sip over static files?

Thanks

Dan Journo
Kesher Communications (UK)
Business Phone Systemshttp://www.keshercommunications.com/ | Hosted 
PBXhttp://www.keshercommunications.com/hostedpbx.html
T: 0161 820 8353


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Re: [asterisk-users] Realtime vs Static Files

[Leandro Dardini]

2013/1/23 Dan Journo d...@keshercommunications.com

 Hi,

 ** **

 We're trying to decide whether to switch back to a static file for
 sip.conf. Currently we use mysql realtime but can't see any real benefit.*
 ***

 ** **

 Why would someone choose realtime sip over static files?

 ** **

 Thanks

 ** **

 Dan Journo

 Kesher Communications (UK)

 Business Phone Systems http://www.keshercommunications.com/ | Hosted 
 PBXhttp://www.keshercommunications.com/hostedpbx.html
 

 T: 0161 820 8353



All depends by the number of sip peers and the number of addition/deletion
you make. If you have static files, you have to sip reload every time you
add/remove a peer. With realtime is all realtime. I have switched to
realtime peers some times ago with great benefit.

Leandro
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Re: [asterisk-users] Realtime vs Static Files

[Dan Journo]

All depends by the number of sip peers and the number of addition/deletion you 
make. If you have static files, you have to sip reload every time you 
add/remove a peer. With realtime is all realtime. I have switched to 
realtime peers some times ago with great benefit.

However, there are limitations with realtime. For example, I've noticed that if 
two asterisk servers share the database, often they both think that the peer is 
registered. Also the issues with MWI and Qualify=

If I automatically generate sip.conf and automatically issue sip reload when 
needed, I can get the same functionality as realtime without the issues 
described above.

Have I missed something?

Thanks
Dan
--
Dan Journo
Kesher Communications (UK)
Business Phone Systemshttp://www.keshercommunications.com/ | Hosted 
PBXhttp://www.keshercommunications.com/hostedpbx.html
T: 0161 820 8353


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Re: [asterisk-users] Realtime vs Static Files

[Ishfaq Malik]

On Wed, 2013-01-23 at 05:53 -0500, Dan Journo wrote:
 For example, I've noticed that if two asterisk servers share the
 database, often they both think that the peer is registered. Also the
 issues with MWI and Qualify=

We have never experienced that and use realtime with multiple asterisk
servers.

The reason we find realtime useful is that you can build your own
interface to interact with the database and pass down peer/dialplan
management to the end user if you are managing a hosted environment.
-- 
Ishfaq Malik i...@pack-net.co.uk
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET
NORTH, MANCHESTER
SCIENCE PARK, MANCHESTER, M156SE
COMPANY REG NO. 04920552


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