Re: [asterisk-users] PJSIP status check at DB level

[Ahmed Munir]

Thanks, by adding contacts in sorcery.conf ps_contacts table is getting
populated however, when getting error below;

ERROR[13270]: res_pjsip_registrar.c:382 register_aor_core: Unable to bind
contact 'sip:20011@10.4.251.25:6275' to AOR '20011'

Currently I've setup Asterisk realtime with res_config_mysql.conf. Is there
a way to fix above error?



Date: Wed, 11 Jan 2017 13:42:36 -0400
> From: Joshua Colp <jc...@digium.com>
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] PJSIP status check at DB level
> Message-ID:
> <1484156556.3940360.844569720.0a814...@webmail.messagingengine.com
> >
> Content-Type: text/plain; charset="utf-8"
>
> On Wed, Jan 11, 2017, at 12:59 PM, Ahmed Munir wrote:
> > Thanks.
> >
> > But I was not able to find the records in 'ps_contacts' table. As per my
> > 'ps_aors' entries, I'm enabling and using only following fields below;
> >
> > id: 20010 
> > max_contacts: 1
> > remove_existing: yes
> > qualify_frequecy: 10
>
> What is your sorcery.conf? Have you configured it such that your
> contacts are put into the database?
>
> --
> 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
>
>
>
>
>


-- 
Regards,

Ahmed Munir Chohan
-- 
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Re: [asterisk-users] PJSIP status check at DB level

[Ahmed Munir]

Thanks.

But I was not able to find the records in 'ps_contacts' table. As per my
'ps_aors' entries, I'm enabling and using only following fields below;

id: 20010 
max_contacts: 1
remove_existing: yes
qualify_frequecy: 10

Please advise if there any more parameters I may need to set in 'ps_aors'
table.


Date: Tue, 10 Jan 2017 10:18:50 -0400
> From: Joshua Colp <jc...@digium.com>
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] PJSIP status check at DB level
> (Realtime)
> Message-ID:
> <1484057930.4162258.843096561.6368d...@webmail.messagingengine.com
> >
> Content-Type: text/plain; charset="utf-8"
>
> On Tue, Jan 10, 2017, at 10:11 AM, Ahmed Munir wrote:
> > Hi,
> >
> > I would like to know how to check PJSIP status for endpoints at DB level
> > (realtime) just like in chan_sip? Like on chan_sip when sip extension
> > gets
> > register regseconds, ipaddr, fullcontact & many more parameters get
> > updated
> > at DB level.
> >
> > The version of asterisk I'm using is 13.8 cert 4
>
> Registered devices are added as contacts to the "ps_contacts" table.
> This is because there can be multiple of them. You would need to look in
> that table and make the association (based on AOR).
>
> --
> 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
>
>


-- 
Regards,

Ahmed Munir Chohan
-- 
_
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[asterisk-users] PJSIP status check at DB level (Realtime)

[Ahmed Munir]

Hi,

I would like to know how to check PJSIP status for endpoints at DB level
(realtime) just like in chan_sip? Like on chan_sip when sip extension gets
register regseconds, ipaddr, fullcontact & many more parameters get updated
at DB level.

The version of asterisk I'm using is 13.8 cert 4

-- 
Regards,

Ahmed Munir Chohan
-- 
_
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Re: [asterisk-users] Openfile Issue

[Ahmed Munir]

See below output;

[root@abc ~]#  lsof -u root | wc -l
5116



From: Dovid Bender <do...@telecurve.com>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users@lists.digium.com>
> Subject: Re: [asterisk-users] Openfile Issue
> Message-ID:
> <CAM3TTh3QA3AqrNNGC1GuSHPGzW1-hGtRPg5hByM1QpBvGcb1Yg@mail.
> gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> 50771 is the PID. I am talking about the user. for instances if running as
> root (which you should never do) then:
>  lsof -u root | wc -l
>
> On Thu, Oct 13, 2016 at 1:31 PM, Ahmed Munir <ahmedmunir...@gmail.com>
> wrote:
>
> >
> > [root@abc asterisk]# lsof -u 50771 | wc -l
> > 0
> >
> > BTW, I'm using CentOS 6.5
> >
> >
> >
> >>
> >> Date: Thu, 13 Oct 2016 10:20:19 -0400
> >>> From: Dovid Bender <do...@telecurve.com>
> >>> To: Asterisk Users Mailing List - Non-Commercial Discussion
> >>> <asterisk-users@lists.digium.com>
> >>> Subject: Re: [asterisk-users] Openfile Issue
> >>> Message-ID:
> >>> 

Re: [asterisk-users] Openfile Issue

[Ahmed Munir]

See below;

[root@abc asterisk]# lsof -u 50771 | wc -l
0

BTW, I'm using CentOS 6.5


> From: Dovid Bender 
>
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> 
>> Subject: Re: [asterisk-users] Openfile Issue
>> Message-ID:
>> 

Re: [asterisk-users] Openfile Issue

[Ahmed Munir]

[root@abc asterisk]# lsof -u 50771 | wc -l
0

BTW, I'm using CentOS 6.5



>
> Date: Thu, 13 Oct 2016 10:20:19 -0400
>> From: Dovid Bender 
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> 
>> Subject: Re: [asterisk-users] Openfile Issue
>> Message-ID:
>> 

Re: [asterisk-users] Openfile Issue

[Ahmed Munir]

See below;

[root@abc asterisk]# cat /proc/50771/limits
Limit Soft Limit   Hard Limit   Units
Max cpu time  unlimitedunlimitedseconds
Max file size unlimitedunlimitedbytes
Max data size unlimitedunlimitedbytes
Max stack size10485760 unlimitedbytes
Max core file sizeunlimitedunlimitedbytes
Max resident set  unlimitedunlimitedbytes
Max processes 256389   256389
processes
Max open files225000files
Max locked memory 6553665536bytes
Max address space unlimitedunlimitedbytes
Max file locksunlimitedunlimitedlocks
Max pending signals   256389   256389   signals
Max msgqueue size 819200   819200   bytes
Max nice priority 00
Max realtime priority 00
Max realtime timeout  unlimitedunlimitedus


Date: Thu, 13 Oct 2016 09:37:34 -0400
> From: Dovid Bender <do...@telecurve.com>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users@lists.digium.com>
> Subject: Re: [asterisk-users] Openfile Issue
> Message-ID:
> <cam3tth2ykzifywjoqf1ezkn9qruvlhli38cmpp6uxjsqck4...@mail.gm
> ail.com>
> Content-Type: text/plain; charset="utf-8"
>
> What do you get when you do:
> cat /proc//limits ?
>
> On Thu, Oct 13, 2016 at 9:30 AM, Ahmed Munir <ahmedmunir...@gmail.com>
> wrote:
>
> > Hi all,
> >
> > Now a days getting openfile issues on asterisk quite often even setting
> > system  soft limit to 2 and hard limit to 25000 and issue usually
> > occurs during openfile socket consumed by system and asterisk is quite
> > smaller than the soft or hard limit. See below system and asterisk logs;
> >
> > 2016:10:13_08:19:01 | Too many LOG file moved successfully - messages
> >
> > 2016:10:13_08:19:01 | Asterisk openfile count: 1252
> >
> > 2016:10:13_08:19:01 | Total system open files count: 4091
> >
> > 2016:10:13_08:19:01 | SIP channels: 93 active calls|1563 calls processed
> >
> > 2016:10:13_08:19:01 | Asterisk SIP peers: 366
> >
> > 2016:10:13_08:19:01 | Asterisk service uptime: System uptime: 4 days, 4
> > hours, 12 minutes, 33 seconds
> >
> > Last reload: 4 days, 4 hours, 12 minutes, 33 seconds
> >
> > Privilege escalation protection disabled!
> >
> > See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
> >
> > 2016:10:13_08:19:01 | Socket Summary
> >
> > Total: 648 (kernel 758)
> >
> > TCP:   20 (estab 8, closed 5, orphaned 0, synrecv 0, timewait 1/0), ports
> > 14
> >
> >
> >
> > Transport Total IPIPv6
> >
> > * 758   - -
> >
> > RAW   0 0 0
> >
> > UDP   422   419   3
> >
> > TCP   15141
> >
> > INET  437   433   4
> >
> > FRAG  0 0 0
> >
> > 2016:10:13_08:19:01 | Logged successfully all the required details
> >
> >
> > [2016-10-13 08:18:41] ERROR[50690][C-062a] res_timing_timerfd.c:
> > Failed to create timerfd timer: Too many open files
> >
> > [2016-10-13 08:18:41] ERROR[50690][C-062a] acl.c: Cannot create
> socket
> >
> > [2016-10-13 08:18:41] ERROR[50690][C-062a] res_timing_timerfd.c:
> > Failed to create timerfd timer: Too many open files
> >
> > [2016-10-13 08:18:41] ERROR[50690][C-062a] acl.c: Cannot create
> socket
> >
> > [2016-10-13 08:18:41] ERROR[50689][C-0629] res_timing_timerfd.c:
> > Failed to create timerfd timer: Too many open files
> >
> > [2016-10-13 08:18:41] ERROR[50689][C-0629] acl.c: Cannot create
> socket
> >
> > [2016-10-13 08:18:40] ERROR[49913][C-05b7] cdr_csv.c: Unable to
> > re-open master file /var/log/asterisk//cdr-csv//Master.csv : Too many
> > open files
> >
> > [2016-10-13 08:18:40] ERROR[2983] acl.c: Cannot create socket
> >
> > [2016-10-13 08:18:40] ERROR[49833][C-059c] cdr_csv.c: Unable to
> > re-open master file /var/log/asterisk//cdr-csv//Master.csv : Too many
> > open files
> >
> > [2016-10-13 08:18:40] ERROR[2983] acl.c: Cannot create socket
> >
> > [2016-10-13 08:18:40] ERROR[2983] acl

[asterisk-users] Openfile Issue

[Ahmed Munir]

Hi all,

Now a days getting openfile issues on asterisk quite often even setting
system  soft limit to 2 and hard limit to 25000 and issue usually
occurs during openfile socket consumed by system and asterisk is quite
smaller than the soft or hard limit. See below system and asterisk logs;

2016:10:13_08:19:01 | Too many LOG file moved successfully - messages

2016:10:13_08:19:01 | Asterisk openfile count: 1252

2016:10:13_08:19:01 | Total system open files count: 4091

2016:10:13_08:19:01 | SIP channels: 93 active calls|1563 calls processed

2016:10:13_08:19:01 | Asterisk SIP peers: 366

2016:10:13_08:19:01 | Asterisk service uptime: System uptime: 4 days, 4
hours, 12 minutes, 33 seconds

Last reload: 4 days, 4 hours, 12 minutes, 33 seconds

Privilege escalation protection disabled!

See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.

2016:10:13_08:19:01 | Socket Summary

Total: 648 (kernel 758)

TCP:   20 (estab 8, closed 5, orphaned 0, synrecv 0, timewait 1/0), ports 14



Transport Total IPIPv6

* 758   - -

RAW   0 0 0

UDP   422   419   3

TCP   15141

INET  437   433   4

FRAG  0 0 0

2016:10:13_08:19:01 | Logged successfully all the required details


[2016-10-13 08:18:41] ERROR[50690][C-062a] res_timing_timerfd.c: Failed
to create timerfd timer: Too many open files

[2016-10-13 08:18:41] ERROR[50690][C-062a] acl.c: Cannot create socket

[2016-10-13 08:18:41] ERROR[50690][C-062a] res_timing_timerfd.c: Failed
to create timerfd timer: Too many open files

[2016-10-13 08:18:41] ERROR[50690][C-062a] acl.c: Cannot create socket

[2016-10-13 08:18:41] ERROR[50689][C-0629] res_timing_timerfd.c: Failed
to create timerfd timer: Too many open files

[2016-10-13 08:18:41] ERROR[50689][C-0629] acl.c: Cannot create socket

[2016-10-13 08:18:40] ERROR[49913][C-05b7] cdr_csv.c: Unable to re-open
master file /var/log/asterisk//cdr-csv//Master.csv : Too many open files

[2016-10-13 08:18:40] ERROR[2983] acl.c: Cannot create socket

[2016-10-13 08:18:40] ERROR[49833][C-059c] cdr_csv.c: Unable to re-open
master file /var/log/asterisk//cdr-csv//Master.csv : Too many open files

[2016-10-13 08:18:40] ERROR[2983] acl.c: Cannot create socket

[2016-10-13 08:18:40] ERROR[2983] acl.c: Cannot create socket

Further added, I'm using CentOS 6.5 as OS.

Please advise what changes required for permanently fixing this random
issue.


-- 
Regards,

Ahmed Munir Chohan
-- 
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Re: [asterisk-users] Asterisk Radius CDR

[Ahmed Munir]

Hi Willy & Andrew,

After doing alot of hits and tries, the issue found at the
dictionary.digium at radius client end. For the solution, I used the
dictionary.digium that comes with asterisk source file, restarted asterisk
services and able to send CDR data over to radius server.

Thanks guys for your help.

On Thu, Sep 29, 2016 at 12:18 PM, Ahmed Munir <ahmedmunir...@gmail.com>
wrote:

> Hi Guys,
>
> Even though enabling Asterisk debug (setting to 9), getting same message
> and not providing enough logs;
>
>  DEBUG[10801][C-]: cdr_radius.c:208 radius_log: Unable to create
> RADIUS record. CDR not recorded!
>
> As per my observation, if I update/rename the dictionary.digium file name
> in dictionary to dictionary.digium1 and unload and load cdr_radius.so
> module, getting message as;
>
>  NOTICE[10792]: cdr_radius.c:271 load_module: Cannot load radiusclient-ng
> dictionary file.
>
> Later I correct it, able to load cdr_radius.so module.
>
> Seems like it there is some issue with cdr_radius.so module itself interm
> of passing data over to radiusclient from asterisk.
>
> Even though I've granted and set permissions 777 for radiusclient configs,
> but the issue remains the same.
>
> Please advise the fix for resolving this issue.
>
>
>
> Date: Thu, 29 Sep 2016 18:11:15 +0800
>> From: Andrew Ivins <and...@ivins.id.au>
>> To: wil...@offermans.rompen.nl
>> Cc: asterisk-users <asterisk-users@lists.digium.com>
>> Subject: Re: [asterisk-users] Asterisk Radius CDR
>> Message-ID:
>> 

Re: [asterisk-users] Asterisk Radius CDR

[Ahmed Munir]

Hi Andrew and Willy,

Thanks for sharing the info.

As for enabling radius server debugging 'radiusd -X', made some test calls
don't see the radiusclient sending data to radius server. However, using
radtest or radiusclient testing, able to send data to radius server (after
enabling debug).

For further testing, on my other server  using OpenSIPs, setup the
radiusclient  and data was able to send over to radius server without any
issue i.e. using same radiusclient config that I'm using for Asterisk
radiusclient.

Btw, will try to work on Andrew advise and will update you if I make any
progress.



Date: Wed, 28 Sep 2016 10:09:51 +0200
> From: Willy Offermans <aster...@offermans.rompen.nl>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users@lists.digium.com>
> Subject: Re: [asterisk-users] Asterisk Radius CDR
> Message-ID: <20160928080951.ga4...@vpn.offrom.nl>
> Content-Type: text/plain; charset=us-ascii
>
> Hello Ahmed, Andrew, and asterisk friends,
>
> Some time ago, I ran into similar problems as well :) I can confirm the
> statement of Andrew: Turn on the logging facilities and you will find your
> issue most likely.  However, you need also a strategy. ``Radius client
> testing'' as you mentioned, can mean anything. The point is, can asterisk
> talk to the freeradius server via the client settings? To my opinion, this
> is easy to test. Maybe the message: ``cdr_radius.c:208 radius_log: Unable
> to create RADIUS record. CDR not recorded'' already implies that this is
> not possible. I cannot judge it. You can by turning on radiusd -X and have
> a close look to the output.
>
> On Wed, Sep 28, 2016 at 07:59:13AM +0800, Andrew Ivins wrote:
> > Hi Ahmed,
> >
> > I ran into similar problems. freeradius-client returns the same error
> code
> > for numerous failure cases, so Asterisk doesn't get an opportunity to log
> > anything useful. If you look here:
> >
> > https://github.com/FreeRADIUS/freeradius-client/blob/master/
> lib/buildreq.c
> >
> > You'll see many instances where it returns ERROR_RC. You are almost
> > certainly running into one of these. I ended up putting in print debug
> into
> > that file and recompiling. I think in my case it was as simple as a
> > hostname not resolving. Once you're not working blind, you'll find what
> is
> > happening pretty quickly.
> >
> > Andrew
> >
> > On 28 September 2016 at 03:32, Ahmed Munir <ahmedmunir...@gmail.com>
> wrote:
> >
> > > I did radius client status testing with radius server, able to access
> the
> > > radius server. However, still getting radius CDR issue after setting
> debug
> > > level 8 even granting 666 access to radiusclient-ng config files.
> > >
> > > message: cdr_radius.c:208 radius_log: Unable to create RADIUS record.
> CDR
> > > not recorded!
> > >
> > > Please advise if I missed out anything.
> > >
> > >
> > > Date: Mon, 26 Sep 2016 12:09:34 +0200
> > >> From: Willy Offermans <aster...@offermans.rompen.nl>
> > >> To: Asterisk Users Mailing List - Non-Commercial Discussion
> > >> <asterisk-users@lists.digium.com>
> > >> Subject: Re: [asterisk-users] Asterisk Radius CDR
> > >> Message-ID: <20160926100934.gb4...@vpn.offrom.nl>
> > >> Content-Type: text/plain; charset=us-ascii
> > >>
> > >>
> > >> Hello Ahmed,
> > >>
> > >> On Fri, Sep 23, 2016 at 04:12:42PM -0400, Ahmed Munir wrote:
> > >> > Hi,
> > >> >
> > >> > I've recently setup Asterisk with Radius CDR by following the
> document:
> > >> > https://wiki.asterisk.org/wiki/display/AST/RADIUS+CDR+Backend.
> > >> >
> > >> > The issue currently I'm facing is after turning on the debug getting
> > >> > message: cdr_radius.c:208 radius_log: Unable to create RADIUS
> record.
> > >> CDR
> > >> > not recorded!
> > >> >
> > >> > I've checked and grant access 666 to radiusclient config files:
> servers
> > >> &
> > >> > dictionary.digium and 777 to '/var/run/radius.seq'. I've noticed
> that
> > >> > /var/run/radius.seq is not getting updated.
> > >> >
> > >> >
> > >> > Further added, in asterisk CLI while running command: cdr show
> status
> > >> > getting results below;
> > >> >
> > >> > Call Detail Record (CDR) settings
> > >> > --
> 

Re: [asterisk-users] Asterisk Radius CDR

[Ahmed Munir]

I did radius client status testing with radius server, able to access the
radius server. However, still getting radius CDR issue after setting debug
level 8 even granting 666 access to radiusclient-ng config files.

message: cdr_radius.c:208 radius_log: Unable to create RADIUS record. CDR
not recorded!

Please advise if I missed out anything.


Date: Mon, 26 Sep 2016 12:09:34 +0200
> From: Willy Offermans <aster...@offermans.rompen.nl>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users@lists.digium.com>
> Subject: Re: [asterisk-users] Asterisk Radius CDR
> Message-ID: <20160926100934.gb4...@vpn.offrom.nl>
> Content-Type: text/plain; charset=us-ascii
>
> Hello Ahmed,
>
> On Fri, Sep 23, 2016 at 04:12:42PM -0400, Ahmed Munir wrote:
> > Hi,
> >
> > I've recently setup Asterisk with Radius CDR by following the document:
> > https://wiki.asterisk.org/wiki/display/AST/RADIUS+CDR+Backend.
> >
> > The issue currently I'm facing is after turning on the debug getting
> > message: cdr_radius.c:208 radius_log: Unable to create RADIUS record. CDR
> > not recorded!
> >
> > I've checked and grant access 666 to radiusclient config files: servers &
> > dictionary.digium and 777 to '/var/run/radius.seq'. I've noticed that
> > /var/run/radius.seq is not getting updated.
> >
> >
> > Further added, in asterisk CLI while running command: cdr show status
> > getting results below;
> >
> > Call Detail Record (CDR) settings
> > --
> >   Logging:Enabled
> >   Mode:   Simple
> >   Log unanswered calls:   No
> >   Log congestion: No
> >
> > * Registered Backends
> >   ---
> > cdr-syslog
> > Adaptive ODBC
> > cdr-custom
> > csv
> > radius
> >
> >
> > Please advise if I may missed any steps.
> >
> > --
> > Regards,
> >
> > Ahmed Munir Chohan
>
> I cannot advice you about steps you might have missed, probably none. To my
> experience, the documentation is not sufficient.
>
> I can tell you that freeradius can be run in debug mode: radiusd -X Do this
> and have a close look to the output.
>
> If you cannot find any attempt to connect to the freeradius server you need
> to have a close look to the asterisk log files as well. Figure out what is
> going wrong. There should be some clue.
>
> I don't understand the grant access settings. Figure out the user which is
> running asterisk and set the setting appropriately! I remember that I
> needed the following access setting:
>
> -rw-r-  1 root  asterisk  /usr/local/etc/radiusclient-ng/servers
>
> So read access for asterisk to the servers file. This was not documented at
> all, but somehow logical, if you figured it out.
>
> --
> Met vriendelijke groeten,
> With kind regards,
> Mit freundlichen Gruessen,
> De jrus wah,
>
> Wiel
>
> *****
>  W.K. Offermans
>
>Powered by 
>
> (__)
>  \\\'',)
>\/  \ ^
>.\._/_)
>
>www.FreeBSD.org
>
>
>
-- 
Regards,

Ahmed Munir Chohan
-- 
_
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[asterisk-users] Asterisk Radius CDR

[Ahmed Munir]

Hi,

I've recently setup Asterisk with Radius CDR by following the document:
https://wiki.asterisk.org/wiki/display/AST/RADIUS+CDR+Backend.

The issue currently I'm facing is after turning on the debug getting
message: cdr_radius.c:208 radius_log: Unable to create RADIUS record. CDR
not recorded!

I've checked and grant access 666 to radiusclient config files: servers &
dictionary.digium and 777 to '/var/run/radius.seq'. I've noticed that
/var/run/radius.seq is not getting updated.


Further added, in asterisk CLI while running command: cdr show status
getting results below;

Call Detail Record (CDR) settings
--
  Logging:Enabled
  Mode:   Simple
  Log unanswered calls:   No
  Log congestion: No

* Registered Backends
  ---
cdr-syslog
Adaptive ODBC
cdr-custom
csv
radius


Please advise if I may missed any steps.

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Ahmed Munir Chohan
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[asterisk-users] Asterisk Realtime RTUPDATE issue

[Ahmed Munir]

Hi,

I'm currently using Asterisk 11.7.0.The issue currently I'm facing in
Asterisk realtime sip_buddies table i.e. if I try to unregister the
extension, ipaddr, port, regseconds, fullcontact, useragent and lastms
remain still populated with data unless do the sip reload. This issue also
obser

In sip.conf the parameter I've enabled/uncommented  for realtime are only
'rtcachefriends=yes' and rest of the realtime parameters are commented (set
as default).

Please advise, what I'm may missed out.

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[asterisk-users] Participant unable to hear other participants in ConfBridge

[Ahmed Munir]

Hi All,

The issue appearing at the random for confbridge module i.e. in some cases
if a participant joins the confbridge, he/she unable to hear others which
make him/her  to hangup the call and redial the bridge again. By joining
the bridge second time, participant able to hear the other participants.

Any ideas which may causing this issue? As Asterisk version I'm using is
11.2.1. Is it a bug? Please advise.



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[asterisk-users] Getting T.38 issue

[Ahmed Munir]

Hi,

Few months back I configured Asterisk 11.6.0 for an outbound fax using T.38
protocol as listing down the flow below;

Asterisk Fax server - (IP) - Cisco VGW -(IP) - Carrier

The issue I'm currently getting when Asterisk receives warnings as listed
below, it is overloading the Cisco VGW, therefore need to restart Asterisk
service or sometimes reboot VGW to clear these warnings.

[Mar 24 09:25:01] WARNING[28645][C-0004] app_fax.c: Unable to write
frame to channel; Resource temporarily unavailable
[Mar 24 09:25:01] WARNING[28613][C-0002] app_fax.c: Unable to write
frame to channel; Resource temporarily unavailable

The configuration in sip.conf for T.38 is listed below;

t38pt_udptl = yes,fec,maxdatagram=400
faxdetect = t38

Udptl.conf;

udptlstart=4000
udptlend=4999
udptlfecentries = 3
udptlfecspan = 3
use_even_ports = no

Please advise at earliest to overcome this issue.

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Re: [asterisk-users] Asterisk Log rotate not working

[Ahmed Munir]

Jim,

Cron and Logrotate already installed in my machine and already configured
as the steps you enlisted. But still logrotate is not running.


Date: Tue, 21 May 2013 12:28:31 -0700
 From: Jim Lucas li...@cmsws.com
 Subject: Re: [asterisk-users] Asterisk Log rotate not working
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Message-ID: 519bcadf.1000...@cmsws.com
 Content-Type: text/plain; charset=ISO-8859-1; format=flowed

 On 5/21/2013 11:54 AM, Ahmed Munir wrote:
  Checked in /var/logs/ directory, all logs are not rotating by logrotate.
  Please advise how can I overcome this issue as I'm using CentoOS 5

 Ahmed,

 Proper log rotation depends on a couple things working together
 correctly to get the job done.  First, you need to make sure you have
 the space to rotate the logs.  If you have compression enabled,
 logrotate creates a copy of the file(s) as it compresses them.  You
 could be running out of space???

 Next you need to verify that everything is in place, follow these steps
 to do so.  Keep in mind that I have CentOS 6.4.  So the packages might
 differ a little in the name and surely in the version numbering.

   1) Verify logrotate is installed to your system.
  # yum install logrotate

  if it asks you to install it, do so.

   2) Verify that crond is installed and running.
  Below is the output I get when searching yum to see if crond is
 installed.  If your query returns nothing then crond is not installed.

[root@jim etc]# yum list all | grep ^cron | grep @
cronie.x86_64 1.4.4-7.el6
 @anaconda-CentOS-201303020151.x86_64/6.4
cronie-anacron.x86_64 1.4.4-7.el6
 @anaconda-CentOS-201303020151.x86_64/6.4
crontabs.noarch   1.10-33.el6
 @anaconda-CentOS-201303020151.x86_64/6.4

  If crond is not installed, then you will need to install it.  Once
 you have it installed, move on to the next step.

   3) Make sure crond is setup to start at boot time.

chkconfig crond on

   4) Verify that logrotate is in one of the cron include folders.  Mine
 is located in the cron.daily folder.

[root@jim etc]# find /etc/*/logrotate
/etc/cron.daily/logrotate

If you don't find that the above file exists, you might need to
 re-install logrotate.

 Next I would've had you verify that you have a config file in
 /etc/logrotate.d/ for the asterisk log files.  But it seems you already
 to.  After all this, if it still isn't working, double check all the
 steps above.

 Let us know if this does or doesn't help.

 --
 Jim Lucas






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[asterisk-users] Asterisk Log rotate not working

[Ahmed Munir]

Hi,

Last year, I installed Asterisk 10.4.2 and enabled logrotate on daily basis
which was working perfect. Now in couple of months back, the logrotate
feature is not working at all but simply appending the logs in 'messages'
file. Listing down down the configuration for logrotate below;

/var/log/asterisk/messages {
missingok
rotate 5
daily
postrotate
/usr/sbin/asterisk -rx 'logger reload'  /dev/null 2 /dev/null
endscript
}

As asterisk is running by user: root so no need set asterisk permissions
'create 0640 asterisk asterisk' in above configuration.

Please advise so I can resolve this issue.



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Re: [asterisk-users] Asterisk Log rotate not working

[Ahmed Munir]

Checked in /var/logs/ directory, all logs are not rotating by logrotate.
Please advise how can I overcome this issue as I'm using CentoOS 5




From: Chris Bagnall aster...@lists.minotaur.cc
 Subject: Re: [asterisk-users] Asterisk Log rotate not working
 To: asterisk-users@lists.digium.com
 Message-ID: 519b9fa6.9000...@lists.minotaur.cc
 Content-Type: text/plain; charset=ISO-8859-1; format=flowed

 On 21/5/13 4:19 pm, Ahmed Munir wrote: Last year, I installed Asterisk
 10.4.2 and enabled logrotate on daily basis
  which was working perfect. Now in couple of months back, the logrotate
  feature is not working at all but simply appending the logs in 'messages'
  file. Listing down down the configuration for logrotate below;

 This sounds more like a Linux/logrotate issue rather than
 asterisk-specific. Are your other system logfiles successfully rotating?
 (e.g. /var/log/messages)

 If not, it may be something as simple as logrotate's daemon not running.
 You should be able to fix that in your distro's startup scripts.

 On Gentoo, you'd do something like /etc/init.d/logrotate start to
 start it now, and rc-update add logrotate default to add it to your
 default runlevel.

 Difficult to advise further without knowing the distro in question.

 Kind regards,

 Chris
 --
 This email is made from 100% recycled electrons





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[asterisk-users] Getting Unknown Error while configuring Asterisk with Linux HA

[Ahmed Munir]

Hi,

I recently configured Linux HA for Asterisk service (using Asterisk
resource agent downloaded from link:
https://github.com/ClusterLabs/resource-agents/blob/master/heartbeat/asterisk
).
As per configuration it is working good but when I include monitor_sipuri=
sip:42@10.3.152.103  parameter in primitive section it is giving me an
errors like listed below;

root@asterisk2 ~ crm_mon -1



Last updated: Thu Mar 28 06:09:54 2013

Stack: Heartbeat

Current DC: asterisk2 (b966dfa2-5973-4dfc-96ba-b2d38319c174) - partition
with quorum

Version: 1.0.12-unknown

2 Nodes configured, unknown expected votes

1 Resources configured.





Online: [ asterisk1 asterisk2 ]



Resource Group: group_1

 asterisk_2 (lsb:asterisk): Started asterisk1

 IPaddr_10_3_152_103(ocf::heartbeat:IPaddr):Started
asterisk1



Failed actions:

p_asterisk_start_0 (node=asterisk1, call=64, rc=1, status=complete):
unknown error

p_asterisk_start_0 (node=asterisk2, call=20, rc=1, status=complete):
unknown error


I tested the 'sipsak' tool on cli, it is executing without any issue i.e.
returning 200 OK but when I remove this param monitor_sipuri I'm not
getting the errors and also I created sip profile '42' without setting any
password, tested first on softphone and is working.

Test result for sipsak;

root@asterisk1 ~ sipsak -v -s sip:42@10.3.152.103
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.3.152.105:60928
;branch=z9hG4bK.274e15e9;alias;received=10.3.152.103;rport=60928
From: sip:sipsak@10.3.152.105:60928;tag=68c5c65d
To: sip:42@10.3.152.103;tag=as558d9271
Call-ID: 1757791837@10.3.152.105
CSeq: 1 OPTIONS
Server: Asterisk PBX 10.12.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Contact: sip:10.3.152.103:5060
Accept: application/sdp
Content-Length: 0


Listing down the configuration below which I configured;

 node $id=887bae58-1eb6-47d1-b539-d12a2ed3d836 asterisk1
node $id=b966dfa2-5973-4dfc-96ba-b2d38319c174 asterisk2
primitive IPaddr_10_3_152_103 ocf:heartbeat:IPaddr \
op monitor interval=5s timeout=20s \
params ip=10.3.152.103
primitive p_asterisk ocf:heartbeat:asterisk \
op monitor interval=10s \
params realtime=true
group group_1 p_asterisk IPaddr_10_3_152_103 \
meta target-role=Started
location rsc_location_group_1 group_1 \
rule $id=preferred_location_group_1 100: #uname eq asterisk1
colocation asterisk-with-ip inf: p_asterisk IPaddr_10_3_152_103
property $id=cib-bootstrap-options \
symmetric-cluster=true \
no-quorum-policy=stop \
default-resource-stickiness=0 \
stonith-enabled=false \
stonith-action=reboot \
startup-fencing=true \
stop-orphan-resources=true \
stop-orphan-actions=true \
remove-after-stop=false \
default-action-timeout=120s \
is-managed-default=true \
cluster-delay=60s \
pe-error-series-max=-1 \
pe-warn-series-max=-1 \
pe-input-series-max=-1 \
dc-version=1.0.12-unknown \
cluster-infrastructure=Heartbeat

And the status I'm getting is listed below;

root@asterisk1 ~ crm_mon -1

Last updated: Fri Mar 29 12:25:10 2013
Stack: Heartbeat
Current DC: asterisk1 (887bae58-1eb6-47d1-b539-d12a2ed3d836) - partition
with quorum
Version: 1.0.12-unknown
2 Nodes configured, unknown expected votes
1 Resources configured.


Online: [ asterisk1 asterisk2 ]

 Resource Group: group_1
 p_asterisk (ocf::heartbeat:asterisk):  Started asterisk1
 IPaddr_10_3_152_103(ocf::heartbeat:IPaddr):Started
asterisk1


Please advise to overcome this issue.


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Regards,

Ahmed Munir Chohan
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Re: [asterisk-users] Getting compilation error while installing Dhadi

[Ahmed Munir]

Thanks Steve and Russ. It worked.


From: Steve Edwards asterisk@sedwards.com
 Subject: Re: [asterisk-users] Getting compilation error while
 installing Dhadi
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Message-ID: alpine.DEB.2.02.1302271337180.3668@ws
 Content-Type: text/plain; charset=iso-8859-1; Format=flowed

 On Wed, 27 Feb 2013, Ahmed Munir wrote:

  I'm getting compilation error as trying to install latest version of
 dahdi

 
 /usr/local/src/Asterisk/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/xpp/xdefs.h:152:
 error: conflicting types for ?bool?
  include/linux/types.h:36: error: previous declaration of ?bool? was here

 Don't let a little thing like a compilation error stop you :)

 Just comment out line 152 in xdefs.h

 There may be a 'proper' way to do this, but this should work.

 I had the same issue compiling zaptel-1.2.27 on CentOS 5.9 yesterday.

 --
 Thanks in advance,
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 Newline  Fax: +1-760-731-3000

 --


 From: Russ Meyerriecks rmeyerrie...@digium.com
 Subject: Re: [asterisk-users] Getting compilation error while
 installing  Dhadi
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Message-ID: 20130227215108.GA17504@blackmagic
 Content-Type: text/plain; charset=iso-8859-1

  error: conflicting types for ?bool?
  include/linux/types.h:36: error: previous declaration of ?bool? was here

 This issue is resolved by the latest dahdi-linux release 2.6.2-rc1.

 You can download a tarball of the release here:

 http://downloads.asterisk.org/pub/telephony/dahdi-linux/dahdi-linux-2.6.2-rc1.tar.gz

 Or you can check out the v2.6.2-rc1 tag from git:
 git clone git.asterisk.org/dahdi/linux dahdi-linux
 cd dahdi-linux
 git checkout v2.6.2-rc1

 --
 Russ Meyerriecks
 Digium, Inc. | Linux Kernel Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 direct: +1 256-428-6025
 Check us out at: www.digium.com  www.asterisk.org


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[asterisk-users] Getting compilation error while installing Dhadi

[Ahmed Munir]

Hi all,

I'm getting compilation error as trying to install latest version of dahdi
on CentOS box 5.9 which I now updated from 5.6. I also installed the
dependencies but still not getting the clue to get install the driver.
Listing down the errors below;


  CC [M]
/usr/local/src/Asterisk/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/wct4xxp/../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_tsi_cnct.o
  CC [M]
/usr/local/src/Asterisk/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/wct4xxp/../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_tsst.o
  CC [M]
/usr/local/src/Asterisk/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/wct4xxp/../oct612x/apilib/bt/octapi_bt0.o
  CC [M]
/usr/local/src/Asterisk/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/wct4xxp/../oct612x/apilib/largmath/octapi_largmath.o
  CC [M]
/usr/local/src/Asterisk/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/wct4xxp/../oct612x/apilib/llman/octapi_llman.o
  LD [M]
/usr/local/src/Asterisk/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/wct4xxp/wct4xxp.o
  CC [M]
/usr/local/src/Asterisk/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/wctc4xxp/base.o
  LD [M]
/usr/local/src/Asterisk/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/wctc4xxp/wctc4xxp.o
  CC [M]
/usr/local/src/Asterisk/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/wctdm24xxp/base.o
  CC [M]
/usr/local/src/Asterisk/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/wctdm24xxp/xhfc.o
  LD [M]
/usr/local/src/Asterisk/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/wctdm24xxp/wctdm24xxp.o
  CC [M]
/usr/local/src/Asterisk/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/wcte12xp/base.o
  LD [M]
/usr/local/src/Asterisk/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/wcte12xp/wcte12xp.o
  VERSION
/usr/local/src/Asterisk/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/xpp/xpp_version.h
  CC [M]
/usr/local/src/Asterisk/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/xpp/card_bri.o
In file included from
/usr/local/src/Asterisk/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/xpp/xpd.h:26,
 from
/usr/local/src/Asterisk/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/xpp/card_bri.c:29:
/usr/local/src/Asterisk/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/xpp/xdefs.h:152:
error: conflicting types for âboolâ
include/linux/types.h:36: error: previous declaration of âboolâ was here
make[4]: ***
[/usr/local/src/Asterisk/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/xpp/card_bri.o]
Error 1
make[3]: ***
[/usr/local/src/Asterisk/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/xpp]
Error 2
make[2]: ***
[_module_/usr/local/src/Asterisk/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi]
Error 2
make[2]: Leaving directory `/usr/src/kernels/2.6.18-348.1.1.el5-x86_64'
make[1]: *** [modules] Error 2
make[1]: Leaving directory
`/usr/local/src/Asterisk/dahdi-linux-complete-2.6.1+2.6.1/linux'
make: *** [all] Error 2


Please advise how can I resolve this issue.



-- 
Regards,

Ahmed Munir Chohan
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[asterisk-users] Configuration Required for Remove Queue Member

[Ahmed Munir]

I would like to know, is there a method in which  we can define the timeout
value for a member who already login to the queue but after quite a while
if he didn't answer the 3-4 calls (not going to member pause queue) but
automatically remove the member from the queue?

Please advise.

-- 
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Re: [asterisk-users] Getting UDPTL (SIP): Transmission error: Resource temporarily unavailable

[Ahmed Munir]

I'm not using the DHCP server configuration and IP addresses  assigned in
the network are manual and there are no clashes found in the network.

The version of Asterisk I'm using is 10.4.2. I think there might be some
issues in this version perhaps I may try to upgrade to 10.12.

UDPTL (SIP/10.3.22.6-0ad6): Transmission error to 10.3.22.6:18428:
Resource temporarily unavailable.

Due to above message, it is badly effecting the V-GW and later I need to
restart the Asterisk service.

Any thoughts on this?


Date: Thu, 17 Jan 2013 15:30:18 +1300
 From: Pete Mundy p...@fiberphone.co.nz
 Subject: Re: [asterisk-users] Getting UDPTL (SIP): Transmission error:

 On 17/01/2013, at 4:35 AM, A J Stiles asterisk_l...@earthshod.co.uk
 wrote:

  Unplug 10.3.22.6, and try pinging it.  If something answers, then you
 indeed
  have a clash.  Check your DHCP server configuration, and make sure any
  manually-assigned addresses are outside its pool of addresses.

 If you do this test, remember to make sure to keep pinging with the host
 disconnected for minimum 30 seconds so as to give your local OS's arp table
 a chance to time out (or manually delete the original ARP entry before
 starting the ping).

 Pete

 --
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Ahmed Munir Chohan
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Re: [asterisk-users] Getting UDPTL (SIP): Transmission error: Resource temporarily unavailable

[Ahmed Munir]

Hi Christopher,

I'm using Asterisk 10.4.2. Do I need to install updated version to resolve
this issue? Please advise.


 --

 Date: Tue, 15 Jan 2013 15:45:31 -0600
 From: Christopher Harrington ch...@acsdi.com
 Subject: Re: [asterisk-users] Getting UDPTL (SIP): Transmission error:
 Resource temporarily unavailable
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Message-ID:
 CAJLBXEnTkUOp=
 ckcfqh2ntzbzxuhu+vsgqvbn+nqc5gytdk...@mail.gmail.com
 Content-Type: text/plain; charset=utf-8

 Can you be more specific about your Asterisk version? 10.xx.yy ?

 Sounds like some sort of resource leak.


 On Tue, Jan 15, 2013 at 3:02 PM, Ahmed Munir ahmedmunir...@gmail.com
 wrote:

  Hi,
 
  I configured Asterisk 10 for inbound fax, for couple of weeks I didn't
 see
  any issues until today. The setup  I configured for inbound fax is quite
  simple i.e. Cisco Voice GW sends the fax calls to Asterisk using T.38
  protocol and later Asterisk stores/forwards the fax to specific end user.
 
  The configuration I made in sip.conf for enabling T38 is listed below;
 
  t38pt_udptl = yes,fec,maxdatagram=400
  faxdetect = t38
 
  And in udptl.conf, I just uncommented 'use_even_ports = yes
  ;' and rest of it set as default.
 
 
  Here is the error I'm usually seeing in Asterisk side;
 
  [Jan 15 14:13:28] NOTICE[20514] udptl.c: UDPTL (SIP/10.3.22.6-0ad6):
  Transmission error to 10.3.22.6:18428: Resource temporarily unavailable
 
  If this notice comes, it occurs repeatedly unless I need to restart the
  asterisk service. For some reason it also effect the V-GW.
 
  Please advise what is the reason that I'm getting this message and how
 can
  I avoid it?
 
 
  --
  Regards,
 
  Ahmed Munir Chohan
 
 
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 Mobile Phone: 612.326.4248
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[asterisk-users] Getting UDPTL (SIP): Transmission error: Resource temporarily unavailable

[Ahmed Munir]

Hi,

I configured Asterisk 10 for inbound fax, for couple of weeks I didn't see
any issues until today. The setup  I configured for inbound fax is quite
simple i.e. Cisco Voice GW sends the fax calls to Asterisk using T.38
protocol and later Asterisk stores/forwards the fax to specific end user.

The configuration I made in sip.conf for enabling T38 is listed below;

t38pt_udptl = yes,fec,maxdatagram=400
faxdetect = t38

And in udptl.conf, I just uncommented 'use_even_ports = yes
;' and rest of it set as default.


Here is the error I'm usually seeing in Asterisk side;

[Jan 15 14:13:28] NOTICE[20514] udptl.c: UDPTL (SIP/10.3.22.6-0ad6):
Transmission error to 10.3.22.6:18428: Resource temporarily unavailable

If this notice comes, it occurs repeatedly unless I need to restart the
asterisk service. For some reason it also effect the V-GW.

Please advise what is the reason that I'm getting this message and how can
I avoid it?


-- 
Regards,

Ahmed Munir Chohan
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[asterisk-users] Storing Custom greeting VM in DB

[Ahmed Munir]

Hi all,

I configured the voicemail using realtime and for record voice messages,
I'm storing it in to MySQL DB as this setup works perfectly without any
issues.  Later I tried to insert the custom greeting (busy) for VM in DB
for particular extension, it was unable to play the custom greeting but
play the default prompt.

Even though, I created the folders (busy and unavail) in the
/var/spool/asterisk/voicemail/default/'1234567 directory, converted the
.wav file to 8KHz 16 bit mono,converted to .gsm format and using default
context for voicemail.  Listing down the data and query;

+--+-+-++--+--+---+---++++-+--++--+--+-+--+---++--+---++-+-+---+---++
| uniqueid | customer_id | context | mailbox| password | fullname |
email | pager | tz | attach | saycid | dialout | callback | review |
operator | envelope | sayduration | saydurationm | sendvoicemail | delete |
nextaftercmd | forcename | forcegreetings | hidefromdir |
stamp   | profile   | forwardno | queue_extn |
+--+-+-++--+--+---+---++++-+--++--+--+-+--+---++--+---++-+-+---+---++
| 8475 | 0   | default | 1234567 |  |  |
|   | en | yes| yes| |  | no | no   |
no   | no  |1 | no| no |
yes  | no| no | yes | 2012-10-12
15:42:40 | voicemail | NULL  | NULL   |
+--+-+-++--+--+---+---++++-+--++--+--+-+--+---++--+---++-+-+---+---+--

INSERT INTO voicemessages
(msgnum,dir,mailboxuser,mailboxcontext,recording)
VALUES
(-1,'/var/spool/asterisk/voicemail/default/'1234567/busy','1234567','default',LOAD_FILE('/var/spool/asterisk/voicemail/default/'1234567/busy.wav')),
(-1,'/var/spool/asterisk/voicemail/default/'1234567/unavail',''1234567','default',LOAD_FILE('/var/spool/asterisk/voicemail/default/'1234567/unavail.wav'));


Do I need to modify any other configuration? Please advise to resolve this
issue.

-- 
Regards,

Ahmed Munir Chohan
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Re: [asterisk-users] Trigger Asterisk after data inserted in MySQL

[Ahmed Munir]

Thanks Bryant and David for sharing.






 From: Bryant Zimmerman brya...@zktech.com
 Subject: Re: [asterisk-users] Trigger Asterisk after data inserted in
 mysql
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Message-ID: 4d5734ed$7d50d85e$77bd1cd6$@zktech.com
 Content-Type: text/plain; charset=us-ascii

 David

 The way we do this is to have a trigger insert into a batch table. This
 table can be polled from a secondary process. That process/service is
 responsible for monitoring, working and cleanup. This allows for you to
 poll a highly optimized table without taking the db performance hit from
 larger tables that will grow over time. We process millions of cdr and
 process records a day this way. It also allows you balanced process loads
 across multiple servers. This can be extremely important on systems that
 are more heavily loaded. It also allows you to remove process load and
 latencies from the database servers.

 Thanks

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

 
  From: David Cook dbc_aster...@advan.ca
 Sent: Wednesday, September 19, 2012 2:04 PM
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Trigger Asterisk after data inserted in
 mysql

 It looks like the answer is yes.


 http://crazytechthoughts.blogspot.ca/2011/12/call-external-program-from-mysq
 l.html

 From the page, here is code to execute a UDF library and call a shell.
 Clearly there would be a heavy penalty to launching a shell so you would
 want to carefully evaluate the frequency this is executed on your system.

 DELIMITER @@   CREATE TRIGGER Test_Trigger  AFTER INSERT ON MyTable  FOR
 EACH ROW  BEGIN  DECLARE cmd CHAR(255);  DECLARE result int(10);  SET
 cmd=CONCAT('sudo /home/sarbac/hello_world ','Sarbajit');  SET result =
 sys_exec(cmd); END; @@ DELIMITER ;

   -dbc

  Message: 1 Date: Tue, 18 Sep 2012 15:41:46 -0400 From: Ahmed Munir
 ahmedmunir...@gmail.com Subject: [asterisk-users] Trigger Asterisk after
 data inserted inmysql To: asterisk-users@lists.digium.comMessage-ID:
 CAGMN=JdbE5FdDSQXxZ9OrWXu3Pvgc-hj-EnPxUrG=
 rjhgsd...@mail.gmail.com
 Content-Type: text/plain; charset=iso-8859-1  Hi all,   I would like to
 know, is there a way to trigger Asterisk after data inserted into mysql DB?
 Like here what I'm trying to do, when the new data inserted into MySQL DB,
 it sends the request to Asterisk along with the new data (that is inserted
 in DB) for making outbound call i.e. Realtime.  Currently I've set a cron
 job that execute my script every 30 seconds and checks for a new data in
 DB. If new data is inserted in 30 seconds that script will run and sends
 the data to Asterisk for making calls. (This is the case which I'm thinking
 to avoid)  Please advise.


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[asterisk-users] Trigger Asterisk after data inserted in mysql

[Ahmed Munir]

Hi all,


I would like to know, is there a way to trigger Asterisk after data
inserted into mysql DB? Like here what I'm trying to do, when the new data
inserted into MySQL DB, it sends the request to Asterisk along with the new
data (that is inserted in DB) for making outbound call i.e. Realtime.

Currently I've set a cron job that execute my script every 30 seconds and
checks for a new data in DB. If new data is inserted in 30 seconds that
script will run and sends the data to Asterisk for making calls. (This is
the case which I'm thinking to avoid)

Please advise.

-- 
Regards,

Ahmed Munir Chohan
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Re: [asterisk-users] Send Fax from Asterisk

[Ahmed Munir]

Thanks for sharing the link. Actually I'm looking for a different approach
without installing/using third party i.e. a user sends an email to Asterisk
(which is also running mail service), as Asterisk receives the mail where
the mail contains attachment and subject contains destination  number,
Asterisk will download the file and capture the number and later send fax
to destination number just like '.call' file.

Does anyone worked on this scenario? If yes/no, please let me know at
earliest.




please check it. might be it will help

 http://ictfax.org/content/installation-guide

 On Tue, Aug 14, 2012 at 7:20 PM, Ahmed Munir ahmedmunir...@gmail.com
 wrote:

  Hi,
 
  I would like to know, anyone who worked in Email to Fax scenario? If so
  please share the idea for implementing it.
 
  As on other hand I configured Asterisk  for inbound Fax which is working
  good i.e. later forward the fax via email but don't know how can I
  implement for outbound fax in this case.
 
  Please advice.
 
  --
  Regards,
 
  Ahmed Munir Chohan
 
 
 Thanks and regards

  Virendra Bhati
 +91-9718500594
 Asterisk Developer
 E-mail-: virbh...@gmail.com
 Skype id:- virbhati2
 New Delhi(India)
 [image: View my profile on
 LinkedIn]http://in.linkedin.com/pub/virendra-bhati/6/a30/755


-- 
Regards,

Ahmed Munir Chohan
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[asterisk-users] Email to Fax solution

[Ahmed Munir]

Hi,

I would like to know, anyone who worked in Email to Fax scenario? If so
please share the idea for implementing it.

As on other hand I configured Asterisk  for inbound Fax which is working
good i.e. later forward the fax via email but don't know how can I
implement for outbound fax in this case.

Please advice.

-- 
Regards,

Ahmed Munir Chohan
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[asterisk-users] SIP client that supports T.38 Fax

[Ahmed Munir]

Hi,

I'm looking for SIP client that supports T.38 Fax other than zoiper.

Please advise at earliest.

-- 
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Re: [asterisk-users] Getting channel 'SIP/192.168.1.69-00000000' refused to negotiate T.38

[Ahmed Munir]

The configuration I did in Cisco Voice GW is listed below;

dial-peer voice 2852 voip
description Incoming Fax Calls to Asterisk
destination-pattern 329..
session protocol sipv2
session target ipv4:192.168.1.69
codec g711ulaw
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none

Please advise to overcome this warning.

On 06/22/2012 12:05 PM, Ahmed Munir wrote:
 
  Here is my setup;
 
  Fax machine - PSTN - Cisco Voice GW - IP cloud - Asterisk. As on
  Cisco Voice GW, T.38 fax already configured on SIP protocol.

 Apparently your configuration of the 'Cisco Voice GW' was not
 successful, as it refused to accept a re-INVITE from Asterisk that
 wanted to switch the SIP channel to T.38 mode.
 --
 Kevin P. Fleming



  Does your VoIP provider support t.38?
 
  Sent from my iPad
 
  On Jun 22, 2012, at 11:05 AM, Ahmed Munir ahmedmunir...@gmail.com
 wrote:
 
   Hi,
  
   I recently configured T.38 on Asterisk 10.4.2. When I send the fax to
  Asterisk, it gives the errors as listed below;
  
WARNING[25986]: app_fax.c:442 transmit_audio: channel
  'SIP/192.168.1.69-' refused to negotiate T.38
WARNING[25986]: app_fax.c:174 span_message: WARNING T.30 ECM carrier
  not found
  
   As in sip.conf the configuration is listed below;
  
   t38pt_udptl = yes,fec,maxdatagram=400
   faxdetect = t38
  
   And the rest are the standard configuration.
  
   Please advise to resolve this issue.
  
   --
   Regards,
  
   Ahmed Munir Chohan
  
 



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[asterisk-users] Getting channel 'SIP/192.168.1.69-00000000' refused to negotiate T.38

[Ahmed Munir]

Hi,

I recently configured T.38 on Asterisk 10.4.2. When I send the fax to
Asterisk, it gives the errors as listed below;

 WARNING[25986]: app_fax.c:442 transmit_audio: channel
'SIP/192.168.1.69-' refused to negotiate T.38
 WARNING[25986]: app_fax.c:174 span_message: WARNING T.30 ECM carrier not
found

As in sip.conf the configuration is listed below;

t38pt_udptl = yes,fec,maxdatagram=400
faxdetect = t38

And the rest are the standard configuration.

Please advise to resolve this issue.

-- 
Regards,

Ahmed Munir Chohan
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Re: [asterisk-users] Getting channel 'SIP/192.168.1.69-00000000' refused to negotiate T.38

[Ahmed Munir]

Here is my setup;

Fax machine - PSTN - Cisco Voice GW - IP cloud - Asterisk. As on Cisco
Voice GW, T.38 fax already configured on SIP protocol.


 Does your VoIP provider support t.38?

 Sent from my iPad

 On Jun 22, 2012, at 11:05 AM, Ahmed Munir ahmedmunir...@gmail.com wrote:

  Hi,
 
  I recently configured T.38 on Asterisk 10.4.2. When I send the fax to
 Asterisk, it gives the errors as listed below;
 
   WARNING[25986]: app_fax.c:442 transmit_audio: channel
 'SIP/192.168.1.69-' refused to negotiate T.38
   WARNING[25986]: app_fax.c:174 span_message: WARNING T.30 ECM carrier
 not found
 
  As in sip.conf the configuration is listed below;
 
  t38pt_udptl = yes,fec,maxdatagram=400
  faxdetect = t38
 
  And the rest are the standard configuration.
 
  Please advise to resolve this issue.
 
  --
  Regards,
 
  Ahmed Munir Chohan
 


-- 
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Ahmed Munir Chohan
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[asterisk-users] Reliable method for FoIP

[Ahmed Munir]

Hi,

I'm looking for a method to setup FoIP i.e. using T.38 protocol with no
PSTN lines.

I tested T.38 feature for Asterisk but the problem I'm getting is unable to
send more than 2 pages but getting timeout error.

Past couple of years I also configured and tested hylafax + iaxmodem for
T.30 faxing but I would like to know whether it  also supports T.38
protocol or not?

Is there any other reliable method available for FoIP? If it is, please
share your views.


-- 
Regards,

Ahmed Munir Chohan
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[asterisk-users] Spandsp supports T.38?

[Ahmed Munir]

Hi,

I would like to know whether SpanDSP supports T.38 for Asterisk 10? Because
as far as using Fax for Asterisk, I'm getting some issues using T.38

-- 
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Ahmed Munir Chohan
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Re: [asterisk-users] asterisk-users Digest, Vol 95, Issue 20

[Ahmed Munir]

Anybody, can you please share your thoughts to overcome this issue?


 Hi,

 I'm getting error: ' FAX session '9' is complete, result: 'FAILED'
 (FAX_FAILURE_PARTIAL), error: '3RD_T2_TIMEOUT', pages: 1, resolution:
 '204x196', transfer rate: '9600', remoteSID: '' ' when I tried send fax
 more than 2 pages to Asterisk using T.38.

 First I set speed rate to 14400 which I was getting same error message
 while sending 2 fax pages document. Later I set the speed rate for sending
 fax machine to 9600(which is the lowest speed rate available fax machine),
 I was able to send 2 pages document fax but tried to send 3 pages document,
 I'm getting this error message.

 The Asterisk version I'm using is 10.4.2. Please advise me at earliest to
 overcome this issue

 Note: Logs can also be provided as per request

 --
 Regards,

 Ahmed Munir Chohan



-- 
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Ahmed Munir Chohan
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[asterisk-users] Getting Error: 3RD_T2_TIMEOUT while using T38 on Asterisk 10

[Ahmed Munir]

Hi,

I'm getting error: ' FAX session '9' is complete, result: 'FAILED'
(FAX_FAILURE_PARTIAL), error: '3RD_T2_TIMEOUT', pages: 1, resolution:
'204x196', transfer rate: '9600', remoteSID: '' ' when I tried send fax
more than 2 pages to Asterisk using T.38.

First I set speed rate to 14400 which I was getting same error message
while sending 2 fax pages document. Later I set the speed rate for sending
fax machine to 9600(which is the lowest speed rate available fax machine),
I was able to send 2 pages document fax but tried to send 3 pages document,
I'm getting this error message.

The Asterisk version I'm using is 10.4.2. Please advise me at earliest to
overcome this issue

Note: Logs can also be provided as per request

-- 
Regards,

Ahmed Munir Chohan
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Re: [asterisk-users] asterisk-users Digest, Vol 95, Issue 6

[Ahmed Munir]

I figured out the problem. Actually the sending fax machine speed was set
as 33000 bps, later I set to 14400 bps and in my dial plan, I forcefully
set to use T.38 protocol. After that I was able to receive fax.

Thanks Tim for assisting me out :).


 - Original Message -

  Hi Tim,

  I'm using Asterisk 10 and on Cisco GW the protocol is set for FAX is
  T.38 and when I try to send the fax from a fax machine i.e. HP 3180,
  I'm getting some warnings as listed below;

  -- Executing [4112345678@default:1] Goto(SIP/192.168.1.69-0005,
  fax-detect,fax,1) in new stack
  -- Goto (fax-detect,fax,1)
  -- Executing [fax@fax-detect:1] NoOp(SIP/192.168.1.69-0005,
   FAX DETECTED ) in new stack
  -- Executing [fax@fax-detect:2] Goto(SIP/192.168.1.69-0005,
  fax-receive,receive,1) in new stack
  -- Goto (fax-receive,receive,1)
  -- Executing [receive@fax-receive:1]
  NoOp(SIP/192.168.1.69-0005,  FAX RECEIVE ) in new
  stack
  -- Executing [receive@fax-receive:2] Set(SIP/192.168.1.69-0005,
  GLOBAL(FAXCOUNT)=5) in new stack
  == Setting global variable 'FAXCOUNT' to '5'
  -- Executing [receive@fax-receive:3] Set(SIP/192.168.1.69-0005,
  FAXCOUNT=5) in new stack
  -- Executing [receive@fax-receive:4] Set(SIP/192.168.1.69-0005,
  FAXFILE=fax-5-rx.tif) in new stack
  -- Executing [receive@fax-receive:5] Set(SIP/192.168.1.69-0005,
  GLOBAL(LASTFAXCALLERNUM)=6461234567) in new stack
  == Setting global variable 'LASTFAXCALLERNUM' to '6461234567'
  -- Executing [receive@fax-receive:6] Set(SIP/192.168.1.69-0005,
  GLOBAL(LASTFAXCALLERNAME)=) in new stack
  == Setting global variable 'LASTFAXCALLERNAME' to ''
  -- Executing [receive@fax-receive:7]
  NoOp(SIP/192.168.1.69-0005,  SETTING FAXOPT ) in new
  stack
  -- Executing [receive@fax-receive:8] Set(SIP/192.168.1.69-0005,
  FAXOPT(ecm)=yes) in new stack
  -- Executing [receive@fax-receive:9] Set(SIP/192.168.1.69-0005,
  FAXOPT(headerinfo)=MY FAXBACK RX) in new stack
  -- Executing [receive@fax-receive:10]
  Set(SIP/192.168.1.69-0005,
  FAXOPT(localstationid)=1234567890) in new stack
  -- Executing [receive@fax-receive:11]
  Set(SIP/192.168.1.69-0005, FAXOPT(maxrate)=14400) in new
  stack
  -- Executing [receive@fax-receive:12]
  Set(SIP/192.168.1.69-0005, FAXOPT(minrate)=2400) in new
  stack
  -- Executing [receive@fax-receive:13]
  NoOp(SIP/192.168.1.69-0005, FAXOPT(ecm) : yes) in new stack
  -- Executing [receive@fax-receive:14]
  NoOp(SIP/192.168.1.69-0005, FAXOPT(headerinfo) : MY FAXBACK
  RX) in new stack
  -- Executing [receive@fax-receive:15]
  NoOp(SIP/192.168.1.69-0005, FAXOPT(localstationid) :
  1234567890) in new stack
  -- Executing [receive@fax-receive:16]
  NoOp(SIP/192.168.1.69-0005, FAXOPT(maxrate) : 14400) in new
  stack
  -- Executing [receive@fax-receive:17]
  NoOp(SIP/192.168.1.69-0005, FAXOPT(minrate) : 2400) in new
  stack
  -- Executing [receive@fax-receive:18]
  NoOp(SIP/192.168.1.69-0005,  RECEIVING FAX : fax-5-rx.tif
  ) in new stack
  -- Executing [receive@fax-receive:19]
  ReceiveFAX(SIP/192.168.1.69-0005,
  /var/spool/asterisk/fax/fax-5-rx.tif) in new stack
  -- Channel 'SIP/192.168.1.69-0005' receiving FAX
  '/var/spool/asterisk/fax/fax-5-rx.tif'
  == Using UDPTL CoS mark 5
  [Jun 4 12:35:02] NOTICE[10371]: chan_sip.c:7577 sip_read: FAX CNG
  detected but no fax extension
  [Jun 4 12:35:02] WARNING[10072]: res_fax.c:1666 receivefax_t38_init:
  channel 'SIP/192.168.1.69-0005' refused to negotiate T.38
  [Jun 4 12:35:02] WARNING[10072]: res_fax.c:1687 receivefax_t38_init:
  Audio FAX not allowed on channel 'SIP/192.168.1.69-0005' and
  T.38 negotiation failed; aborting.
  [Jun 4 12:35:02] ERROR[10072]: res_fax.c:1891 receivefax_exec: error
  initializing channel 'SIP/192.168.1.69-0005' in T.38 mode
  == Spawn extension (fax-receive, receive, 19) exited non-zero on
  'SIP/192.168.1.69-0005'

  In my sip.conf global configuration I enabled 'fax detect' and
  't38pt_udptl' and added Cisco VGW peer;

  [CiscoVGW-10.70.X.X]
  host=10.70.X.X
  type=friend
  disallow=all
  allow=ulaw
  allow=alaw
  nat=yes
  insecure=port,invite
  context=fax-call
  canreinvite=no
  qualify=yes
  dtmfmode=inband


 T.38 failed to negotiate. That means either your Asterisk side, or your
 Cisco side are not playing nicely together. A packet capture of the call
 setup would be helpful to determine which side is having the issues.

 --Tim

 --
Regards,

Ahmed Munir Chohan
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Re: [asterisk-users] Fax over IP ?

[Ahmed Munir]

On Tue, Jun 5, 2012 at 12:47 PM, Ahmed Munir ahmedmunir...@gmail.comwrote:

 I figured out the problem. Actually the sending fax machine speed was set
 as 33000 bps, later I set to 14400 bps and in my dial plan, I forcefully
 set to use T.38 protocol. After that I was able to receive fax.

 Thanks Tim for assisting me out :).



 - Original Message -

  Hi Tim,

  I'm using Asterisk 10 and on Cisco GW the protocol is set for FAX is
  T.38 and when I try to send the fax from a fax machine i.e. HP 3180,
  I'm getting some warnings as listed below;

  -- Executing [4112345678@default:1] Goto(SIP/192.168.1.69-0005,
  fax-detect,fax,1) in new stack
  -- Goto (fax-detect,fax,1)
  -- Executing [fax@fax-detect:1] NoOp(SIP/192.168.1.69-0005,
   FAX DETECTED ) in new stack
  -- Executing [fax@fax-detect:2] Goto(SIP/192.168.1.69-0005,
  fax-receive,receive,1) in new stack
  -- Goto (fax-receive,receive,1)
  -- Executing [receive@fax-receive:1]
  NoOp(SIP/192.168.1.69-0005,  FAX RECEIVE ) in new
  stack
  -- Executing [receive@fax-receive:2] Set(SIP/192.168.1.69-0005,
  GLOBAL(FAXCOUNT)=5) in new stack
  == Setting global variable 'FAXCOUNT' to '5'
  -- Executing [receive@fax-receive:3] Set(SIP/192.168.1.69-0005,
  FAXCOUNT=5) in new stack
  -- Executing [receive@fax-receive:4] Set(SIP/192.168.1.69-0005,
  FAXFILE=fax-5-rx.tif) in new stack
  -- Executing [receive@fax-receive:5] Set(SIP/192.168.1.69-0005,
  GLOBAL(LASTFAXCALLERNUM)=6461234567) in new stack
  == Setting global variable 'LASTFAXCALLERNUM' to '6461234567'
  -- Executing [receive@fax-receive:6] Set(SIP/192.168.1.69-0005,
  GLOBAL(LASTFAXCALLERNAME)=) in new stack
  == Setting global variable 'LASTFAXCALLERNAME' to ''
  -- Executing [receive@fax-receive:7]
  NoOp(SIP/192.168.1.69-0005,  SETTING FAXOPT ) in new
  stack
  -- Executing [receive@fax-receive:8] Set(SIP/192.168.1.69-0005,
  FAXOPT(ecm)=yes) in new stack
  -- Executing [receive@fax-receive:9] Set(SIP/192.168.1.69-0005,
  FAXOPT(headerinfo)=MY FAXBACK RX) in new stack
  -- Executing [receive@fax-receive:10]
  Set(SIP/192.168.1.69-0005,
  FAXOPT(localstationid)=1234567890) in new stack
  -- Executing [receive@fax-receive:11]
  Set(SIP/192.168.1.69-0005, FAXOPT(maxrate)=14400) in new
  stack
  -- Executing [receive@fax-receive:12]
  Set(SIP/192.168.1.69-0005, FAXOPT(minrate)=2400) in new
  stack
  -- Executing [receive@fax-receive:13]
  NoOp(SIP/192.168.1.69-0005, FAXOPT(ecm) : yes) in new stack
  -- Executing [receive@fax-receive:14]
  NoOp(SIP/192.168.1.69-0005, FAXOPT(headerinfo) : MY FAXBACK
  RX) in new stack
  -- Executing [receive@fax-receive:15]
  NoOp(SIP/192.168.1.69-0005, FAXOPT(localstationid) :
  1234567890) in new stack
  -- Executing [receive@fax-receive:16]
  NoOp(SIP/192.168.1.69-0005, FAXOPT(maxrate) : 14400) in new
  stack
  -- Executing [receive@fax-receive:17]
  NoOp(SIP/192.168.1.69-0005, FAXOPT(minrate) : 2400) in new
  stack
  -- Executing [receive@fax-receive:18]
  NoOp(SIP/192.168.1.69-0005,  RECEIVING FAX : fax-5-rx.tif
  ) in new stack
  -- Executing [receive@fax-receive:19]
  ReceiveFAX(SIP/192.168.1.69-0005,
  /var/spool/asterisk/fax/fax-5-rx.tif) in new stack
  -- Channel 'SIP/192.168.1.69-0005' receiving FAX
  '/var/spool/asterisk/fax/fax-5-rx.tif'
  == Using UDPTL CoS mark 5
  [Jun 4 12:35:02] NOTICE[10371]: chan_sip.c:7577 sip_read: FAX CNG
  detected but no fax extension
  [Jun 4 12:35:02] WARNING[10072]: res_fax.c:1666 receivefax_t38_init:
  channel 'SIP/192.168.1.69-0005' refused to negotiate T.38
  [Jun 4 12:35:02] WARNING[10072]: res_fax.c:1687 receivefax_t38_init:
  Audio FAX not allowed on channel 'SIP/192.168.1.69-0005' and
  T.38 negotiation failed; aborting.
  [Jun 4 12:35:02] ERROR[10072]: res_fax.c:1891 receivefax_exec: error
  initializing channel 'SIP/192.168.1.69-0005' in T.38 mode
  == Spawn extension (fax-receive, receive, 19) exited non-zero on
  'SIP/192.168.1.69-0005'

  In my sip.conf global configuration I enabled 'fax detect' and
  't38pt_udptl' and added Cisco VGW peer;

  [CiscoVGW-10.70.X.X]
  host=10.70.X.X
  type=friend
  disallow=all
  allow=ulaw
  allow=alaw
  nat=yes
  insecure=port,invite
  context=fax-call
  canreinvite=no
  qualify=yes
  dtmfmode=inband


 T.38 failed to negotiate. That means either your Asterisk side, or your
 Cisco side are not playing nicely together. A packet capture of the call
 setup would be helpful to determine which side is having the issues.

 --Tim

 --
 Regards,

 Ahmed Munir Chohan





-- 
Regards,

Ahmed Munir Chohan
--
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Re: [asterisk-users] Fax over IP?

[Ahmed Munir]

Hi Tim,

I'm using Asterisk 10 and on Cisco GW the protocol is set for FAX is T.38
and when I try to send the fax from a fax machine i.e. HP 3180, I'm getting
some warnings as listed below;

-- Executing [4112345678@default:1] Goto(SIP/192.168.1.69-0005,
fax-detect,fax,1) in new stack
-- Goto (fax-detect,fax,1)
-- Executing [fax@fax-detect:1] NoOp(SIP/192.168.1.69-0005, 
FAX DETECTED ) in new stack
-- Executing [fax@fax-detect:2] Goto(SIP/192.168.1.69-0005,
fax-receive,receive,1) in new stack
-- Goto (fax-receive,receive,1)
-- Executing [receive@fax-receive:1] NoOp(SIP/192.168.1.69-0005,
 FAX RECEIVE ) in new stack
-- Executing [receive@fax-receive:2] Set(SIP/192.168.1.69-0005,
GLOBAL(FAXCOUNT)=5) in new stack
  == Setting global variable 'FAXCOUNT' to '5'
-- Executing [receive@fax-receive:3] Set(SIP/192.168.1.69-0005,
FAXCOUNT=5) in new stack
-- Executing [receive@fax-receive:4] Set(SIP/192.168.1.69-0005,
FAXFILE=fax-5-rx.tif) in new stack
-- Executing [receive@fax-receive:5] Set(SIP/192.168.1.69-0005,
GLOBAL(LASTFAXCALLERNUM)=6461234567) in new stack
  == Setting global variable 'LASTFAXCALLERNUM' to '6461234567'
-- Executing [receive@fax-receive:6] Set(SIP/192.168.1.69-0005,
GLOBAL(LASTFAXCALLERNAME)=) in new stack
  == Setting global variable 'LASTFAXCALLERNAME' to ''
-- Executing [receive@fax-receive:7] NoOp(SIP/192.168.1.69-0005,
 SETTING FAXOPT ) in new stack
-- Executing [receive@fax-receive:8] Set(SIP/192.168.1.69-0005,
FAXOPT(ecm)=yes) in new stack
-- Executing [receive@fax-receive:9] Set(SIP/192.168.1.69-0005,
FAXOPT(headerinfo)=MY FAXBACK RX) in new stack
-- Executing [receive@fax-receive:10] Set(SIP/192.168.1.69-0005,
FAXOPT(localstationid)=1234567890) in new stack
-- Executing [receive@fax-receive:11] Set(SIP/192.168.1.69-0005,
FAXOPT(maxrate)=14400) in new stack
-- Executing [receive@fax-receive:12] Set(SIP/192.168.1.69-0005,
FAXOPT(minrate)=2400) in new stack
-- Executing [receive@fax-receive:13] NoOp(SIP/192.168.1.69-0005,
FAXOPT(ecm) : yes) in new stack
-- Executing [receive@fax-receive:14] NoOp(SIP/192.168.1.69-0005,
FAXOPT(headerinfo) : MY FAXBACK RX) in new stack
-- Executing [receive@fax-receive:15] NoOp(SIP/192.168.1.69-0005,
FAXOPT(localstationid) : 1234567890) in new stack
-- Executing [receive@fax-receive:16] NoOp(SIP/192.168.1.69-0005,
FAXOPT(maxrate) : 14400) in new stack
-- Executing [receive@fax-receive:17] NoOp(SIP/192.168.1.69-0005,
FAXOPT(minrate) : 2400) in new stack
-- Executing [receive@fax-receive:18] NoOp(SIP/192.168.1.69-0005,
 RECEIVING FAX : fax-5-rx.tif ) in new stack
-- Executing [receive@fax-receive:19]
ReceiveFAX(SIP/192.168.1.69-0005,
/var/spool/asterisk/fax/fax-5-rx.tif) in new stack
-- Channel 'SIP/192.168.1.69-0005' receiving FAX
'/var/spool/asterisk/fax/fax-5-rx.tif'
  == Using UDPTL CoS mark 5
[Jun  4 12:35:02] NOTICE[10371]: chan_sip.c:7577 sip_read: FAX CNG detected
but no fax extension
[Jun  4 12:35:02] WARNING[10072]: res_fax.c:1666 receivefax_t38_init:
channel 'SIP/192.168.1.69-0005' refused to negotiate T.38
[Jun  4 12:35:02] WARNING[10072]: res_fax.c:1687 receivefax_t38_init: Audio
FAX not allowed on channel 'SIP/192.168.1.69-0005' and T.38 negotiation
failed; aborting.
[Jun  4 12:35:02] ERROR[10072]: res_fax.c:1891 receivefax_exec: error
initializing channel 'SIP/192.168.1.69-0005' in T.38 mode
  == Spawn extension (fax-receive, receive, 19) exited non-zero on
'SIP/192.168.1.69-0005'

In my sip.conf global configuration I enabled 'fax detect' and
't38pt_udptl' and added Cisco VGW peer;

[CiscoVGW-10.70.X.X]
host=10.70.X.X
type=friend
disallow=all
allow=ulaw
allow=alaw
nat=yes
insecure=port,invite
context=fax-call
canreinvite=no
qualify=yes
dtmfmode=inband


While the fax machine starts to send the fax after a while it gives the
message, 'Fax failed' with error code: '388'. Is it the end point fax
machine issue or else? Please assist me out to resolve this issue at
earliest.




  Thanks for your response. Here is my topology as listing down below;

  PSTN Line -- Cisco Voice GW -- IP Cloud -- Asterisk

  Will Asterisk able to receive the fax (as in topology above) using
  its' fax module? In sip.conf I enabled fax detection and T.38.
  Actually I don't want
  to use Hylafax + iaxmodem as per requirement.

 If your Cisco voice gateway can deliver the calls using T.38, that should
 give you decent reliability. You'll want to us Asterisk 10 which has the
 best T.38 support at this point (compared to older releases). The receiving
 side of the equation then becomes whether to use Fax for Asterisk
 (commercial, 1 free channel, 2+ paid), or the included SpanDSP based fax
 module.

 --Tim


-- 
Regards,

Ahmed Munir Chohan
--
_
-- Bandwidth

[asterisk-users] Fax over IP ?

[Ahmed Munir]

Hi all,

Couple of things I would like ask, does Asterisk provides free license for
FoIP (for 1 channel) or need to purchase it? Couple of years back, I was
able to send and receive the fax using Digium T1 card, in term of FoIP how
can I able to receive fax from traditional telephone lines / T1 lines? As
far my understanding, the functionality for FoIP is to send fax to email or
receive fax from email i.e. using T.38 protocol.

The thing I would like to know how I can implement this solution i.e.
receiving fax via IP? Correct me if I'm wrong, while receiving fax from
traditional telephone lines will the topology looks like as listed below;

PSTN Lines -- Asterisk (mounted a T1/ analog card) -- IP -- Asterisk
(receive Fax over IP)

or else?

Please advice.


-- 
Regards,

Ahmed Munir Chohan
--
_
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[asterisk-users] Getting Remote UNIX connection disconnected

[Ahmed Munir]

Hi,

 I'm getting the messages listed below after login to asterisk cli;
   -- Remote UNIX connection
   -- Remote UNIX connection disconnected

   Usually verbose level is set to 4, after setting to 2, I'm not getting
these messages.

   Is there other way to stop these messages? because I'm getting very
irritated and need to set verbose level at least 4.

   Further added, I also tried to stop and start asterisk service but still
getting these messages.
   --
   Regards,
   Ahmed Munir Chohan

-- 
Regards,

Ahmed Munir Chohan
--
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[asterisk-users] Getting Ulimit Message after restart asterisk service

[Ahmed Munir]

Hi all,

Currently I'm getting this message after restarting asterisk service;

 Starting asterisk: /usr/sbin/safe_asterisk: line 86: ulimit: open files:
cannot modify limit: Operation not permitted

Before when I had root access I was not facing this message after that
system administrator assigned me sudo access for restarting asterisk
service.

Please assist me out to resolve this issue at earliest. I also tried to set
ulimit till 4-40 times higher than currently set i.e. 1024 but still giving
me same message.


-- 
Regards,

Ahmed Munir Chohan
--
_
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[asterisk-users] Getting Ulimit Message after restart asterisk service

[Ahmed Munir]

Hi all,

Currently I'm getting this message after restarting asterisk service;

 Starting asterisk: /usr/sbin/safe_asterisk: line 86: ulimit: open files:
cannot modify limit: Operation not permitted

Before when I had root access I was not facing this message after that
system administrator assigned me sudo access for restarting asterisk
service.

Please assist me out to resolve this issue at earliest. I also tried to set
ulimit till 4-40 times higher than currently set i.e. 1024 but still giving
me same message.


-- 
Regards,

Ahmed Munir Chohan
--
_
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Re: [asterisk-users] Getting Ulimit Message after restart asterisk service

[Ahmed Munir]

Thanks Danny,

I would like to know do I need to worry about this message? And why I'm
getting this ulimit message? Please provide reason briefly


From: Danny Nicholas da...@debsinc.com

 Subject: Re: [asterisk-users] Getting Ulimit Message after restart
asteriskservice
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
 Message-ID: 00c101ccf6f2$9e8c11c0$dba43540$@debsinc.com
 Content-Type: text/plain; charset=us-ascii

 This one is simple.  Open /usr/sbin/safe_asterisk and put # in first
 character of line 86 and 102.  Or modfy /etc/sudoers to allow your sudo to
 execute ulimit.



 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir
 Sent: Wednesday, February 29, 2012 8:52 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Getting Ulimit Message after restart asterisk
 service



 Hi all,

 Currently I'm getting this message after restarting asterisk service;

  Starting asterisk: /usr/sbin/safe_asterisk: line 86: ulimit: open files:
 cannot modify limit: Operation not permitted

 Before when I had root access I was not facing this message after that
 system administrator assigned me sudo access for restarting asterisk
 service.

 Please assist me out to resolve this issue at earliest. I also tried to set
 ulimit till 4-40 times higher than currently set i.e. 1024 but still giving
 me same message.


 --
 Regards,

 Ahmed Munir Chohan



-- 
Regards,

Ahmed Munir Chohan
--
_
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Re: [asterisk-users] Getting one way audio even NAT is configured

[Ahmed Munir]

Hi Warren,

Device A is behind NAT with regards to asterisk server. As far as localnet
statement first I did configured localnet = 130.8.2.0/255.255.255.0 as per
local network, after that made a SIP call and the message I'm getting is
listed below;

[Feb  2 11:14:52] WARNING[23868]: chan_sip.c:3622 retrans_pkt:
Retransmission timeout reached on transmission
OGEzODA0MzI4MWE1NzdiZDlkNmQ3NjYyMzJjYzYyOTY. for seqno 1 (Critical
Response) -- See
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[Feb  2 11:14:52] WARNING[23868]: chan_sip.c:3651 retrans_pkt: Hanging up
call OGEzODA0MzI4MWE1NzdiZDlkNmQ3NjYyMzJjYzYyOTY. - no reply to our
critical packet (see
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

So after setting to 130.0.0.0/130.0.0.0 I wasn't getting the above warning
message but facing one way audio.

Date: Wed, 1 Feb 2012 14:38:01 -0600
 From: Warren Selby wcse...@selbytech.com
 Subject: Re: [asterisk-users] Getting one way audio even NAT is
configured
 To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
 Message-ID:
CAM_w8OJmX0nXfdU06p=-fprabz2h7tqr-mjmjnfnweavkjs...@mail.gmail.com
 
 Content-Type: text/plain; charset=iso-8859-1

 On Wed, Feb 1, 2012 at 1:16 PM, Ahmed Munir ahmedmunir...@gmail.com
 wrote:

  Hi all,
 
  I'm getting one way audio when calling over the SIP trunk i.e. end device
  B (remote end of SIP trunk) can hear device A (softphone registered with
  Asterisk) but device A can't hear device B. Even though I configured same
  NAT configurations on other servers and they are working good. The NAT
  configuration is listed below;
 
  localnet=130.0.0.0/130.0.0.0
  externhost=12.131.12.13
  externrefresh=10
  fromdomain=test.localhost.com
  nat=yes
  qualify=yes
  canreinvite=no
 
 
  NAT on device end i.e. my softphone (extension) has already set to yes
  with canreinvite=no  but still unable to resolve this issue. SIP traces
 are
  listed below;
 
 
 snip


 
  The Asterisk version I'm using is 1.8.5. Please assist me at earliest.
 

 Which device (A or B) is behind NAT with regards to your asterisk server?
 Is that the actual localnet= statement you're using, because to my
 understanding that is not the proper format to use (should be
 localnet=x.x.x.x/y.y.y.y where x.x.x.x is your actual local network, and
 y.y.y.y is your subnet for your local network).

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


--
Regards,

Ahmed Munir Chohan
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[asterisk-users] Getting one way audio even NAT is configured

[Ahmed Munir]

-0024
-- Locally bridging SIP/2005-0024 and
SIP/ATTLABS-IP-FlexReach-0025
Reliably Transmitting (NAT) to 12.194.12.12:5060:
OPTIONS sip:12.194.12.12 SIP/2.0
Via: SIP/2.0/UDP 12.131.12.13:5060;branch=z9hG4bK06532068;rport
Max-Forwards: 70
From: Unknown sip:unkn...@test.localhost.com;tag=as054a7d2d
To: sip:12.194.12.12
Contact: sip:Unknown@12.131.12.13:5060
Call-ID: 767dcb7d4406d06c248a7056559ad...@test.localhost.com
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.8.5.0)
Date: Wed, 01 Feb 2012 16:11:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0


---

--- SIP read from UDP:12.194.12.12:5060 ---
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 12.131.12.13:5060
;received=12.131.12.13;branch=z9hG4bK06532068;rport=5060
From: Unknown sip:unkn...@test.localhost.com;tag=as054a7d2d
To: sip:12.194.12.12;tag=aprqngfrt-d1v40r1c6
Call-ID: 767dcb7d4406d06c248a7056559ad...@test.localhost.com
CSeq: 102 OPTIONS
Reason: Q.850;cause=55;text=Call Terminated
Allow: INVITE,ACK,BYE,CANCEL,PRACK,INFO,REFER,UPDATE,MESSAGE,PUBLISH

-
--- (8 headers 0 lines) ---
Scheduling destruction of SIP dialog '
04ce1d566f1f17a221caba261e2af...@test.localhost.com' in 6400 ms (Method:
INVITE)
set_destination: Parsing sip:12.194.12.12:5060;transport=udp for
address/port to send to
set_destination: set destination to 12.194.12.12:5060
Reliably Transmitting (NAT) to 12.194.12.12:5060:
BYE sip:12.194.12.12:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 12.131.12.13:5060;branch=z9hG4bK2ab85b31;rport
Max-Forwards: 70
From: 77057 sip:77...@test.localhost.com;tag=as1fa9b502
To: sip:173242@12.194.12.12
;tag=SDk39gc99-7026517720142726_c2b08.2.2.1323416803184.0_1194082_2371400
Call-ID: 04ce1d566f1f17a221caba261e2af...@test.localhost.com
CSeq: 103 BYE
User-Agent: FPBX-2.9.0(1.8.5.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
--- SIP read from UDP:12.194.12.12:5060 ---
SIP/2.0 200 OK
Via: SIP/2.0/UDP 12.131.12.13:5060
;received=12.131.12.13;branch=z9hG4bK2ab85b31;rport=5060
From: 77057 sip:77...@test.localhost.com;tag=as1fa9b502
To: sip:173242@12.194.12.12
;tag=SDk39gc99-7026517720142726_c2b08.2.2.1323416803184.0_1194082_2371400
Call-ID: 04ce1d566f1f17a221caba261e2af...@test.localhost.com
CSeq: 103 BYE
Content-Length: 0

-
--- (7 headers 0 lines) ---
Really destroying SIP dialog '
04ce1d566f1f17a221caba261e2af...@test.localhost.com' Method: INVITE


The Asterisk version I'm using is 1.8.5. Please assist me at earliest.

-- 
Regards,

Ahmed Munir Chohan
--
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Re: [asterisk-users] asterisk - AGI (perl) - sqlplus(oracle)

[Ahmed Munir]

Yes, I already declared 'use lib
/home/asterisk/lib/lib64/perl5/5.8.8/x86_64-linux-thread-multi/;' in my
AGI. When I execute the script as a user Asterisk, i.e. perl -wc test.pl in
return I'm getting OK and no error messages and script is running fine when
I try to run in shell.

Even though I already declared the environmental variables in
.bash_profile. At the end I tired every method but still stuck in this
problem.



Date: Thu, 5 Jan 2012 14:07:59 -0800
 From: Ron Bergin r...@i.frys.com
 Subject: Re: [asterisk-users] asterisk - AGI (perl) - sqlplus
(oracle)
 To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
 Message-ID:
69ecf8ff3230bc206837478422f97aad.squir...@webmail.i.frys.com
 Content-Type: text/plain;charset=iso-8859-1

 Ahmed Munir wrote:
  Hi,
 
  I installed the modules in asterisk user home directory with read and
  excitable permissions for asterisk but still my AGI not working.

 IMO, it would have been better to install it in it's normal location.

 Is your script using the warnings and strict pragmas?

 What error message do you receive when running the script from the command
 line?

 Did you add the proper use lib '' statement to add the install
 directory to the @INC array?

 Ron Bergin

 
  Please provide me other advise to resolve this issue.
 
 
  Date: Wed, 4 Jan 2012 11:30:34 -0600
  From: Danny Nicholas da...@debsinc.com
  Subject: Re: [asterisk-users] asterisk - AGI (perl) - sqlplus
 (oracle)
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 asterisk-users@lists.digium.com
  Message-ID: 00ca01cccb06$911e8300$b35b8900$@debsinc.com
  Content-Type: text/plain; charset=us-ascii
 
  The module probably isn't readable/executeable from Asterisk
 
 
 
  From: asterisk-users-boun...@lists.digium.com
  [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed
  Munir
  Sent: Wednesday, January 04, 2012 10:45 AM
  To: asterisk-users@lists.digium.com
  Subject: [asterisk-users] asterisk - AGI (perl) - sqlplus (oracle)
 
 
 
  Hi all,
 
  I'm trying to run an AGI in PERL which uses the module DBD-Oracle.
  Currently
  my AGI is working fine in my two servers but not in my other four
  servers.
  When  I tried execute an AGI (as a user asterisk) in command line it
  works
  fine (even I also declare environmental variables in user profile and in
  my
  AGI), but when I tried to call my AGI (perl) in dial plan, it don't get
  executed.
 
  Please advise me to resolve this issue.
 
  --
  Regards,
 
  Ahmed Munir Chohan
 
 




-- 
Regards,

Ahmed Munir Chohan
--
_
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Re: [asterisk-users] asterisk - AGI (perl) - sqlplus(oracle)

[Ahmed Munir]

The thing is, my AGI is working fine if I don't include DBD::Oracle library
in my script. If I include DBD::Oracle library my AGI script gets aborted.
I installed DBD::Oracle module in asterisk application home directory as
its' permissions are listed below;

[asterisk@klpi062 ~]$ ls -lh
/home/asterisk/perl-lib/lib/lib64/perl5/site_perl/5.8.8/x86_64-linux-thread-multi/
total 40K
drwxr-xr-x 3 asterisk asterisk 4.0K Jan  3 14:16 auto
drwxr-xr-x 3 asterisk asterisk 4.0K Jan  3 14:16 DBD
-rwxr-xr-x 1 asterisk asterisk 1.3K Aug 26 14:09 oraperl.ph
-rwxr-xr-x 1 asterisk asterisk  28K Oct 12 12:43 Oraperl.pm

I also included the library path for locating DBD::Oracle module in my AGI.
But still unable to understand even though asterisk has permissions to
access DBD module but still AGI don't work when I include DBD library in it.



 Date: Wed, 4 Jan 2012 12:18:24 -0600
 From: Danny Nicholas da...@debsinc.com
 Subject: Re: [asterisk-users] asterisk - AGI (perl) - sqlplus
(oracle)
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
 Message-ID: 010101cccb0d$3fa6a140$bef3e3c0$@debsinc.com
 Content-Type: text/plain; charset=us-ascii

 What are the permissions on the module you are trying to run? (ls -l
 /var/lib/asterisk/agi-bin/module)



 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir
 Sent: Wednesday, January 04, 2012 12:15 PM
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] asterisk - AGI (perl) - sqlplus (oracle)



 Hi,

 I installed the modules in asterisk user home directory with read and
 excitable permissions for asterisk but still my AGI not working.

 Please provide me other advise to resolve this issue.


 Date: Wed, 4 Jan 2012 11:30:34 -0600
 From: Danny Nicholas da...@debsinc.com
 Subject: Re: [asterisk-users] asterisk - AGI (perl) - sqlplus
   (oracle)
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
   asterisk-users@lists.digium.com
 Message-ID: 00ca01cccb06$911e8300$b35b8900$@debsinc.com
 Content-Type: text/plain; charset=us-ascii

 The module probably isn't readable/executeable from Asterisk



 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir
 Sent: Wednesday, January 04, 2012 10:45 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] asterisk - AGI (perl) - sqlplus (oracle)



 Hi all,

 I'm trying to run an AGI in PERL which uses the module DBD-Oracle.
 Currently
 my AGI is working fine in my two servers but not in my other four servers.
 When  I tried execute an AGI (as a user asterisk) in command line it works
 fine (even I also declare environmental variables in user profile and in my
 AGI), but when I tried to call my AGI (perl) in dial plan, it don't get
 executed.

 Please advise me to resolve this issue.

 --
 Regards,

 Ahmed Munir Chohan




 --
Regards,

Ahmed Munir Chohan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] asterisk - AGI (perl) - sqlplus (oracle)

[Ahmed Munir]

Hi all,

I'm trying to run an AGI in PERL which uses the module DBD-Oracle.
Currently my AGI is working fine in my two servers but not in my other four
servers. When  I tried execute an AGI (as a user asterisk) in command line
it works fine (even I also declare environmental variables in user profile
and in my AGI), but when I tried to call my AGI (perl) in dial plan, it
don't get executed.

Please advise me to resolve this issue.

-- 
Regards,

Ahmed Munir Chohan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] asterisk - AGI (perl) - sqlplus (oracle)

[Ahmed Munir]

Hi,

I installed the modules in asterisk user home directory with read and
excitable permissions for asterisk but still my AGI not working.

Please provide me other advise to resolve this issue.


 Date: Wed, 4 Jan 2012 11:30:34 -0600
 From: Danny Nicholas da...@debsinc.com
 Subject: Re: [asterisk-users] asterisk - AGI (perl) - sqlplus
(oracle)
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
 Message-ID: 00ca01cccb06$911e8300$b35b8900$@debsinc.com
 Content-Type: text/plain; charset=us-ascii

 The module probably isn't readable/executeable from Asterisk



 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir
 Sent: Wednesday, January 04, 2012 10:45 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] asterisk - AGI (perl) - sqlplus (oracle)



 Hi all,

 I'm trying to run an AGI in PERL which uses the module DBD-Oracle.
 Currently
 my AGI is working fine in my two servers but not in my other four servers.
 When  I tried execute an AGI (as a user asterisk) in command line it works
 fine (even I also declare environmental variables in user profile and in my
 AGI), but when I tried to call my AGI (perl) in dial plan, it don't get
 executed.

 Please advise me to resolve this issue.

 --
 Regards,

 Ahmed Munir Chohan


 -
Regards,

Ahmed Munir Chohan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Check which client access Asterisk using AMI

[Ahmed Munir]

Hi,

In manager.conf file I created a user profile by which clients can access
Asterisk server as listed below;


[cbusapp]
secret = cbus123
deny=0.0.0.0/0.0.0.0
permit=192.168.1.0/255.255.255.0
read = system,call,log,verbose,command,agent,user,originate
write = system,call,log,verbose,command,agent,user,originate



Using above configuration clients are successfully access the asterisk and
forward its parameters to asterisk. The thing I would like to know how can I
keep track from which client does asterisk receives request from? Like
client A, B and C need to know from which clients the request was made to
asterisk.

-- 
Regards,

Ahmed Munir Chohan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Check which client access Asterisk using AMI

[Ahmed Munir]

Is there other way around doing it instead of enabling debug and verbose
from logger.conf?





 Message: 1
 Date: Thu, 27 Oct 2011 10:36:08 -0400
 From: Ahmed Munir ahmedmunir...@gmail.com
 Subject: [asterisk-users] Check which client access Asterisk using AMI
 To: asterisk-users@lists.digium.com
 Message-ID:
CAGMN=JdNtgAu-yjWB_-Yi7rr=0jk5osjqz7ibygufgejd4b...@mail.gmail.com
 
 Content-Type: text/plain; charset=iso-8859-1

 Hi,

 In manager.conf file I created a user profile by which clients can access
 Asterisk server as listed below;


 [cbusapp]
 secret = cbus123
 deny=0.0.0.0/0.0.0.0
 permit=192.168.1.0/255.255.255.0
 read = system,call,log,verbose,command,agent,user,originate
 write = system,call,log,verbose,command,agent,user,originate



 Using above configuration clients are successfully access the asterisk and
 forward its parameters to asterisk. The thing I would like to know how can
 I
 keep track from which client does asterisk receives request from? Like
 client A, B and C need to know from which clients the request was made to
 asterisk.

 --
 Regards,

 Ahmed Munir Chohan
 -- next part --
 An HTML attachment was scrubbed...
 URL: 
 http://lists.digium.com/pipermail/asterisk-users/attachments/20111027/ff89ea1a/attachment.html
 

 --

 Message: 2
 Date: Thu, 27 Oct 2011 09:39:43 -0500
 From: Danny Nicholas da...@debsinc.com
 Subject: Re: [asterisk-users] Check which client access Asterisk using
AMI
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
 Message-ID: 006e01cc94b6$443a6c60$ccaf4520$@debsinc.com
 Content-Type: text/plain; charset=us-ascii

 This information might be in /var/log/asterisk/messages or /v/l/a/full.  If
 not, you can change the logging and get it there (turn on debug in one of
 them) (/etc/asterisk/logger.conf)



 --
Regards,

Ahmed Munir Chohan
--
_
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[asterisk-users] sending fax using chan_capi

[Ahmed Munir]

Hi,

I tried to sendfax a text file, it was received successfully and the context
were in ascii format (readable form). As I tried to send a fax in .tiff
format (converted from pdf format using ghostscript), the context I received
in fax is in binary form. The dial plan is listed below;

exten = 100,1,Verbose( Sending Dialogic Diva Fax...)
exten = 100,n,set(BeforeFaxTime=${EPOCH})
exten = 100,n,capicommand(sendfax,/tmp/out.tiff,732-XXX-,Dialogic Diva
Test Sendfax)
exten = 100,n,HangUp()
exten = h,1,set(ElapsedFaxTime=$[${EPOCH}-${BeforeFaxTime}])
exten =
h,n,AGI(printfaxresults.sh,${FAXSTATUS},${FAXREASON},${FAXREASONTEXT},${FAXRATE},${FAXRESOLUTION},${FAXFORMAT},${FAXCFFFORMAT},${FAXPAGES},${FAXID},${FAXEXTEN},${ElapsedFaxTime},FaxesSent.log)




Please advice, how can I send fax in image format using T.30

-- 
Regards,

Ahmed Munir Chohan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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