[asterisk-users] Queues - how to add back a agent without all other calls to agents stoping and re-starting

2020-07-23 Thread Paddy Grice 
Hi All

 

I have a problem with queues that I have been trying to solve for many
months - the customer has now picked back up onto this and wanting a
solution - any guidance, ideas or solutions welcome.

 

This is the situation :-

 

We have a number of agents in a ringall group, a call joins the queue and
all handsets ring - great. 

An agent answers and handles the call - great.

 

The call is cleared but the agents handset is not re-rung (for the next
call) until timeout seconds elapse at that time all other agents handsets
stop ringing for a moment and then start again.

 

As I say any help or guidance appreciated

 

Paddy

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Re: [asterisk-users] Length of dial string

2020-05-04 Thread Paddy Grice 
Thanks for the info - will build latest version and test 

Paddy


-Original Message-
From: Floimair Florian [mailto:f.floim...@commend.com] 
Sent: 04 May 2020 08:33
To: pa...@wizaner.com; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Length of dial string

Hi Paddy!

This used to be 80 characters total (including all characters like channel
type, '&' and '/'. Had the same issue in the past where I extended that in
the code and recompiled.
From what I understand there is basically no longer a hard limit in Dial
since the recent change in the latest versions other than a single device
must not exceed this but you can concatenate any number of them within the
Dial string.

Hi all

as per the new release notice for 13.33.0 received today - can anyone
advise
me the max limit of the string to the Dial Command - see 
*   [ASTERISK-27946
 
https://eur01.safelinks.protection.outlook.com/?url=https%3A%2F%2F
issues.asterisk.org%2Fjira%2Fbrowse%2FASTERISK-27946data=02%7C01%7Cf.fl
oimair%40commend.com%7C5a5c413f7d8747dab6c408d7eda07497%7C13b1ddb756454e7fbe
663171548559da%7C0%7C0%7C637239145718403259sdata=JdT9Yvi7ml%2FqzIYMO39k
s68rdMKY2P2DFIAGKCCh6a8%3Dreserved=0> ] - 
dial (API): Storage of dialed target uses AST_MAX_EXTENSION
when it shouldn't

I have been fighting with this issue for months trying to find a
solution I
need to call 20+ devices at the same time so dial strings are very long
I
cant really use a queue(ringall) which was my original idea as the
customer
needs different groups for virtually every call some of which are simple
sip
devices and others have to be local devices (Internal and External
CLIs). 

    Paddy Grice





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Re: [asterisk-users] Length of dial string

2020-05-01 Thread Paddy Grice 
Hi Dovid
 
Yes was one of the options but as the required list is dynamic becomes very
messy - and all combinations problem - where as "call all workers job xxx"
is what is needed so the ability to call 20+ numbers is what is needed - agi
does a database search for all jobx workers and constructs a dialstring with
SIP, DAHDI and Local devices. 
 
Can someone tell me where to find maximum string length for the dial data in
the DIAL command 
 
Paddy
 
  _  

From: Dovid Bender [mailto:do...@telecurve.com] 
Sent: 01 May 2020 10:26
To: pa...@wizaner.com; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Length of dial string


Paddy, 

Why not use local extensions? You can do something like this.
Exten =>
s,1,Dial(Local/set1@call_all/set2@call_all/set3@call_all)

[call_all]
Exten => set1,1,Dial(SIP/100/101/102/103/104/105
Exten => set1,1,Dial(SIP/106/107/108/109/110/111
Exten => set1,1,Dial(SIP/112/113/114/1015/116/117


On Fri, May 1, 2020 at 3:22 AM Paddy Grice  wrote:


Hi all

as per the new release notice for 13.33.0 received today - can anyone advise
me the max limit of the string to the Dial Command - see 
*   [ASTERISK-27946
https://issues.asterisk.org/jira/browse/ASTERISK-27946> ] - 
dial (API): Storage of dialed target uses AST_MAX_EXTENSION
when it shouldn't

I have been fighting with this issue for months trying to find a solution I
need to call 20+ devices at the same time so dial strings are very long I
cant really use a queue(ringall) which was my original idea as the customer
needs different groups for virtually every call some of which are simple sip
devices and others have to be local devices (Internal and External CLIs). 

Paddy Grice





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[asterisk-users] Length of dial string

2020-05-01 Thread Paddy Grice 
Hi all
 
as per the new release notice for 13.33.0 received today - can anyone advise
me the max limit of the string to the Dial Command - see 
*   [ASTERISK-27946
https://issues.asterisk.org/jira/browse/ASTERISK-27946> ] - 
dial (API): Storage of dialed target uses AST_MAX_EXTENSION
when it shouldn't

I have been fighting with this issue for months trying to find a solution I
need to call 20+ devices at the same time so dial strings are very long I
cant really use a queue(ringall) which was my original idea as the customer
needs different groups for virtually every call some of which are simple sip
devices and others have to be local devices (Internal and External CLIs). 

Paddy Grice


 


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Re: [asterisk-users] PJSIP Lockup

2020-04-01 Thread Paddy Grice 
Hi All
 
This sounds just like a problem I have had and still investigating having
moved to 16.9 using chan_sip. I am still trying to repeat the problem it
looks from debug that the issue is either voicemail of call transfer but I
cant consistently repeat it. 
 
Voicemail is using ODBC and I just imported the data from the old system
into the new database.
 
Nick - if you have any more info I would be grateful 
 
TIA
 
Paddy
 
  _  

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On
Behalf Of Nick Olsen
Sent: 01 April 2020 18:54
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP Lockup


We ultimately found this to be a voicemail issue. The voicemail is held in
MYSQL as well (via ODBC). And we found when attempting to playback a
customers voicemail unavail greeting is when the deadlock would occur
(Immediately, every time. Throwing the same "task processors" errors, And
making pjsip completely unresponsive). We had imported a number of greetings
from a legacy asterisk system and the vast majority of them worked. When we
deleted the row containing the customers unavail greeting (making asterisk
revert to read the mailbox number) all issues went away. If we re-record the
customers unavail greeting it works fine and the problem doesn't reoccur.
This was one out of ~250 voicemails imported. 

Since then we've done a few more migrations and they've all gone smooth with
the exception of the most recent one. ~50% of the imported greetings have
caused asterisk to deadlock. We've been checking them now at time of
migration.

What I can't figure out is what it doesn't like about the greeting. It was
on a previous asterisk system working fine. The row looks identical to a
working one. The only thing I can guess is something about the blob for the
recording goes wrong. It would be nice if asterisk handled that more
gracefully. 

I post this mostly just for internet history. To hopefully help the next guy
out who has this same issue.

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



On Mon, Mar 2, 2020 at 8:29 PM Joshua C. Colp  wrote:


On Mon, Mar 2, 2020 at 4:24 PM Nick Olsen 
wrote:


Thanks for the info, Joshua. 

Does PJSIP handle database access the same way Chan_sip did? We had a number
of boxes running chan_sip referencing the same mysql server without issue.

We're going to attempt to get a backtrace on the next occurance. We're also
going to run a local copy of the database on the same physical asterisk
instance and have the system reference it. Just to "throw everything at the
wall".


It uses the same underlying API and layer. It can do more frequent database
access though due to queries and because PJSIP is multithreaded.

-- 

Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org

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Re: [asterisk-users] Queues and penalties

2018-11-30 Thread Paddy Grice 
Thanks Leon 
 
I will implement and test but I knew there would be a fix for what I believe
is a short coming in app_queue. How do I suggest this as a option to the
base code?
 
Paddy

  _  

From: Leon Wright [mailto:lwri...@corpcloud.com.au] 
Sent: 30 November 2018 02:17
To: pa...@wizaner.com; asterisk-users@lists.digium.com
Cc: johnkinis...@gmail.com
Subject: Re: [asterisk-users] Queues and penalties


Paddy, 

This appears to be how the queue app works. I ended up patching the queue
app:

diff --git a/apps/app_queue.c b/apps/app_queue.c
index e3a4e22..72072d0 100644
--- a/apps/app_queue.c
+++ b/apps/app_queue.c
@@ -4571,7 +4571,7 @@ static int ring_one(struct queue_ent *qe, struct
callattempt *outgoing, int *bus
struct callattempt *cur;
/* Ring everyone who shares this best metric (for
ringall) */
for (cur = outgoing; cur; cur = cur->q_next) {
-   if (cur->stillgoing && !cur->chan &&
cur->metric <= best->metric) {
+   if (cur->stillgoing && !cur->chan &&
cur->metric >= qe->min_penalty * 100 && cur->metric <= qe->max_penalty *
100) {
ast_debug(1, "(Parallel) Trying '%s'
with metric %d\n", cur->interface, cur->metric);
ret |= ring_entry(qe, cur, busies);
}

So the penalties get calculated during the 'ringall' strategy and allowing
the queue app to exit, looping and raising the max penalty and calling the
queue app again.

Leon

On Thu, 29 Nov 2018 at 18:24, Paddy Grice  wrote:



Hi John
 
This works fine providing extensions 1001,1002 and 1003 are "Incall" or
"Paused" - the problem appears to be that is a handset say 1002 is "ringing"
then the 2xxx then the penalty is not honoured.
 
This is well described in the History section of the following link
https://wiki.freepbx.org/display/PPS/lazymembers+patch+to+app_queue
 
As I say this seems to be a real shortcoming in app_queue. 
 
Any ideas, suggestions, anyone want to work with me to sort this ?
 
Paddy
 
 
  _  

From: John Kiniston [mailto:johnkinis...@gmail.com] 
Sent: 28 November 2018 21:17
To: pa...@wizaner.com; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Queues and penalties


This should work, How are you defining your timeouts in the queues.conf ? 


And to verify, in your extensions.conf you are calling Queue with the queue
name and the ruleset to apply from queuerules.conf? 


On Wed, Nov 28, 2018 at 12:45 PM Paddy Grice  wrote:



Hi All 
 
I have been looking at this problem for a few days/weeks now and after some
advice please.
 
I currently have a customer on 11.25.3 and I am in the process of upgrading
versions and OS (Debian) and all things that involves mysql -> PDO etc
 
The problem I have is the customer want a simple call distribution like this
 
Extn 1001, 1002, 1003 to be called on an incoming call - if they don't
answer after 20 seconds then 2001, 2002, 2003 to be added to the ringing
extensions and if no one answers after another 20 seconds the add in 3001,
3002, 3003.
 
Seems a simple queue application to me
 
1001, 1002 and 1003 in the queue with a penalty of 1 strategy ringall

2001, 2002 and 2003 in the queue with a penalty of 2 strategy ringall

3001, 3002 and 3003 in the queue with a penalty of 3 strategy ringall
 
and rules 
 
increasing the maxpenalty 1->2 after 20 seconds
and increasing maxpenalty 2->3 after another 20 seconds.
 
But this doesn't work if users don't answer!!
 
if user 1002 or (2001 etc)  just lets his phone ring - he forgot to logoff
or DND then the penalty is ignored.
 
There seems to have been a patch for FreePBX on V13 - LazyMembers - but that
is all I can find and later versions have no mention of this 
 
I guess I can use autopause and some AMI / Script but this stops phones
ringing because of the timeout so the user has a ringing phone and then it
stops and then it starts again whereas the penalty just adds handsets into
the ringing group.
 
This seems to be a real shortcoming in app_queue.
 
Any ideas, suggestions, anyone want to work with me to sort this ?
 
Paddy Grice
 
 
 
 
 

 
 
 
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Re: [asterisk-users] Queues and penalties

2018-11-29 Thread Paddy Grice 
Hi John
 
This works fine providing extensions 1001,1002 and 1003 are "Incall" or
"Paused" - the problem appears to be that is a handset say 1002 is "ringing"
then the 2xxx then the penalty is not honoured.
 
This is well described in the History section of the following link
https://wiki.freepbx.org/display/PPS/lazymembers+patch+to+app_queue
 
As I say this seems to be a real shortcoming in app_queue. 
 
Any ideas, suggestions, anyone want to work with me to sort this ?
 
Paddy
 
 
  _  

From: John Kiniston [mailto:johnkinis...@gmail.com] 
Sent: 28 November 2018 21:17
To: pa...@wizaner.com; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Queues and penalties


This should work, How are you defining your timeouts in the queues.conf ? 


And to verify, in your extensions.conf you are calling Queue with the queue
name and the ruleset to apply from queuerules.conf? 


On Wed, Nov 28, 2018 at 12:45 PM Paddy Grice  wrote:



Hi All 
 
I have been looking at this problem for a few days/weeks now and after some
advice please.
 
I currently have a customer on 11.25.3 and I am in the process of upgrading
versions and OS (Debian) and all things that involves mysql -> PDO etc
 
The problem I have is the customer want a simple call distribution like this
 
Extn 1001, 1002, 1003 to be called on an incoming call - if they don't
answer after 20 seconds then 2001, 2002, 2003 to be added to the ringing
extensions and if no one answers after another 20 seconds the add in 3001,
3002, 3003.
 
Seems a simple queue application to me
 
1001, 1002 and 1003 in the queue with a penalty of 1 strategy ringall

2001, 2002 and 2003 in the queue with a penalty of 2 strategy ringall

3001, 3002 and 3003 in the queue with a penalty of 3 strategy ringall
 
and rules 
 
increasing the maxpenalty 1->2 after 20 seconds
and increasing maxpenalty 2->3 after another 20 seconds.
 
But this doesn't work if users don't answer!!
 
if user 1002 or (2001 etc)  just lets his phone ring - he forgot to logoff
or DND then the penalty is ignored.
 
There seems to have been a patch for FreePBX on V13 - LazyMembers - but that
is all I can find and later versions have no mention of this 
 
I guess I can use autopause and some AMI / Script but this stops phones
ringing because of the timeout so the user has a ringing phone and then it
stops and then it starts again whereas the penalty just adds handsets into
the ringing group.
 
This seems to be a real shortcoming in app_queue.
 
Any ideas, suggestions, anyone want to work with me to sort this ?
 
Paddy Grice
 
 
 
 
 

 
 
 
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[asterisk-users] Queues and penalties

2018-11-28 Thread Paddy Grice 
Hi All 
 
I have been looking at this problem for a few days/weeks now and after some
advice please.
 
I currently have a customer on 11.25.3 and I am in the process of upgrading
versions and OS (Debian) and all things that involves mysql -> PDO etc
 
The problem I have is the customer want a simple call distribution like this
 
Extn 1001, 1002, 1003 to be called on an incoming call - if they don't
answer after 20 seconds then 2001, 2002, 2003 to be added to the ringing
extensions and if no one answers after another 20 seconds the add in 3001,
3002, 3003.
 
Seems a simple queue application to me
 
1001, 1002 and 1003 in the queue with a penalty of 1 strategy ringall
2001, 2002 and 2003 in the queue with a penalty of 2 strategy ringall
3001, 3002 and 3003 in the queue with a penalty of 3 strategy ringall
 
and rules 
 
increasing the maxpenalty 1->2 after 20 seconds
and increasing maxpenalty 2->3 after another 20 seconds.
 
But this doesn't work if users don't answer!!
 
if user 1002 or (2001 etc)  just lets his phone ring - he forgot to logoff
or DND then the penalty is ignored.
 
There seems to have been a patch for FreePBX on V13 - LazyMembers - but that
is all I can find and later versions have no mention of this 
 
I guess I can use autopause and some AMI / Script but this stops phones
ringing because of the timeout so the user has a ringing phone and then it
stops and then it starts again whereas the penalty just adds handsets into
the ringing group.
 
This seems to be a real shortcoming in app_queue.
 
Any ideas, suggestions, anyone want to work with me to sort this ?
 
Paddy Grice
 
 
 
 
 
 
 
 
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Re: [asterisk-users] Asterisk removes a charachter from sip peer name

2015-01-05 Thread Paddy Grice 
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olli Heiskanen
Sent: 03 January 2015 08:04
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk removes a charachter from sip peer name






Hello all,


Just wondering on a behavior I noticed while testing with realtime sip peers
with names like 111@mydomain.com. Using Kamailio as outbound proxy, it
sends Asterisk a sip message where To header value is
sip:111@mydomain.com mailto:sip%3a111@mydomain.com  and From
header has value username sip:111@mydomain.com
mailto:sip%3a111@mydomain.com ;transport=UDP;tag=fc609171. When
Asterisk sends out the sip message, the To header is as it was but as for
From header, Asterisk removes the . charachter from the user part of the
sip uri, thus resulting in 111333. Also the Contact header is affected the
same way.


I was wondering what might be causing this? Does Asterisk not allow dots in
the peer names? The call itself connects so it's not much of an issue but it
would be good to know about this, as of course there's a chance I've just
missed something relevant.


cheers,
Olli 
 
Sounds a bit like  
 
From sip.conf
 
; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and
'-' not
; in square brackets.  For example, the caller id value 555. becomes
555
; when this option is enabled.  Disabling this option results in no
modification
; of the caller id value, which is necessary when the caller id represents
something
; that must be preserved.  This option can only be used in the [general]
section.
; By default this option is on.
;
;shrinkcallerid=yes ; on by default

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

2014-12-16 Thread Paddy Grice 
Hi All
 
I am trying to get voicemail switched over to ARA on version 13 and notice
that the password is not stored in the db when it is changed.
 
A little hair pulling and playing around and I think the problem is in the
function ast_update2_realtime in main/config.c.
 
Issued source is ==
 
int ast_update2_realtime(const char *family, ...)
{
RAII_VAR(struct ast_variable *, lookup_fields, NULL,
ast_variables_destroy);
RAII_VAR(struct ast_variable *, update_fields, NULL,
ast_variables_destroy);
va_list ap;
 
va_start(ap, family);
/* XXX: If we wanted to pass no lookup fields (select all), we'd be
 * out of luck. realtime_arguments_to_fields expects at least one
key
 * value pair. */
realtime_arguments_to_fields(ap, lookup_fields);
va_end(ap);
 
va_start(ap, family);
realtime_arguments_to_fields2(ap, 1, lookup_fields);
va_end(ap);
 
if (!lookup_fields || !update_fields) {
return -1;
}
 
return ast_update2_realtime_fields(family, lookup_fields,
update_fields);
}
 
I believe line 3314 of the file main/config.c should be 
 
realtime_arguments_to_fields2(ap, 1, update_fields);
 
I have changed it and it works for me - but - 
 
1)I don't know what else this may effect 
2)I dont know how to pass this on to the development team
 
Paddy
 
 
 
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Re: [asterisk-users] 'restart when convenient'

2014-05-28 Thread Paddy Grice 
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Larsen
Sent: 28 May 2014 16:47
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 'restart when convenient'


asterisk-users-boun...@lists.digium.com wrote on 05/28/2014 10:37:25 AM:

 pbx1*CLI core restart when convenient 
 Waiting for inactivity to perform restart
 Ignoring asterisk restart request, already in progress.
 
 
 After doing 'core restart now' and hitting Enter really hard ;) Asterisk
 did restart.
 
 
 Some how Asterisk thinks it is not convenient. I want to find out why.
 

I haven't had it fail to restart, but I have been in the same situation and
had it have a nice delay of a minute or two before it finally finds it
convenient to restart. Haven't figured out what the delay is though. I am on
11.6.0. 
 
Maybe nothing but I had a similar problem with ... when convenient - seem
to remember it was a problem writing a cdr using cdr_adaptive_odbc - a
database fault. Asterisk appeared idle but wouldn't close as I guess it
thought something was still active. 
 
Paddy 
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[asterisk-users] Looking for Email to Fax Solutions

2011-06-08 Thread Paddy Grice 
Hi All
 
I am looking for a small scale Email to fax solution 
 
Searches seem to throw up 
 
AsterFax http://sourceforge.net/projects/asterfax/ which seems to go to
http://www.noojee.com.au/products/noojee-fax/fax-overview/
email12fax http://wpkg.org/email2fax/index.php/Main_Page
 
I would appreciate any comments on these or other solutions
 
I am running asterisk 1.4 and I am looking for a small scale solution say 10
lines (ddis)
 
Thanks in advance
 
Paddy
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Re: [asterisk-users] Looking for Email to Fax Solutions

2011-06-08 Thread Paddy Grice 
On 06/08/2011 01:09 AM, Paddy Grice wrote:
 Hi All
  
 I am looking for a small scale Email to fax solution
  
 Searches seem to throw up
  
 AsterFax http://sourceforge.net/projects/asterfax/ which seems to go 
 to http://www.noojee.com.au/products/noojee-fax/fax-overview/
 email12fax http://wpkg.org/email2fax/index.php/Main_Page
  
 I would appreciate any comments on these or other solutions
  
 I am running asterisk 1.4 and I am looking for a small scale solution 
 say 10 lines (ddis)

While I designed it with Asterisk 1.6 or 1.8 in mind, you may try this:

http://messinet.com/trac/wiki/AsteriskFAXGateway

I have some time next week if it needs some tweaks to work with Asterisk
1.4.  -A

--
Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E

Looks like it will certainly help me - I will work through it and let you
know - I am away for a few days so will be next week before I can try it
out. 

Thanks

Paddy


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[asterisk-users] CLI - displaying all channel variables

2011-05-11 Thread Paddy Grice 
Hi List
 
This may be a silly question by web searches etc don't seem to answer it. 
 
Is there a CLI command to display ALL channel variables - standard and user
created - for a specific channel?
 
something like show channel SIP/Test123 all
 
I'm using Version 1.4.33.1
 
PG
 
 
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Re: [asterisk-users] CLI - displaying all channel variables

2011-05-11 Thread Paddy Grice 
Thanks Danny - That displays user created variables - by user this could
be application like dial but not the predefined channel variables.



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: 11 May 2011 17:03
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] CLI - displaying all channel variables

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- 
 boun...@lists.digium.com] On Behalf Of Eric Wieling
 Sent: Wednesday, May 11, 2011 10:55 AM
 To: pa...@wizaner.com; Asterisk Users Mailing List -Non-Commercial 
 Discussion
 Subject: Re: [asterisk-users] CLI - displaying all channel variables
 
 
 
  -Original Message-
  From: asterisk-users-boun...@lists.digium.com
  [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paddy 
  Grice
  Sent: Wednesday, May 11, 2011 11:49 AM
  To: asterisk-users@lists.digium.com
  Subject: [asterisk-users] CLI - displaying all channel variables
 
  Hi List
 
  This may be a silly question by web searches etc don't seem to 
  answer it.
 
  Is there a CLI command to display ALL channel variables - standard 
  and user created - for a specific channel?
 
  something like show channel SIP/Test123 all
 
 The dialplan application DumpChan dumps information about the channel, 
 however, it does not display all the variables you are looking for.
 Generally you should insert a Noop in the dialplan to examine variables.
 Noop(EXTEN is ${EXTEN}) for example.
 
[Danny Nicholas]
Try core show channel sip/test123-001 - you will probably have to do a
core show channels first to get the proper -001 value.




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[asterisk-users] Asterisk -rx command not returning data - Version 1.4.33.1

2011-03-14 Thread Paddy Grice 
Hi List
 
I am having trouble running the command 
 
siptest:~# asterisk -rx 'dialplan reload'

most times it does what I expect and I get a response as below
 
siptest:~# asterisk -rx 'dialplan reload'
Dialplan reloaded.
 
every now and then I get no response i.e.
 
siptest:~# asterisk -rx 'dialplan reload'
siptest:~# 
 
and a verbose 10 setting shows 
 
[Mar 14 19:07:41] ERROR[3092]: utils.c:968 ast_carefulwrite: write()
returned error: Broken pipe
-- Remote UNIX connection disconnected
siptest*CLI 
 
I assume the problem is timing but any ideas on how to fix it 
 
The test is running in an idle box Debian Lenny asterisk version 1.4.33.1
unpatched
 
Any ideas would be helpful
 
Paddy
 
 
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Re: [asterisk-users] Asterisk -rx command not returning data - Version 1.4.33.1

2011-03-14 Thread Paddy Grice 
Hi Steve

I agree - past experience has been hit and miss for me too - I have to do a
dialplan reload after an external database update and have been advised
against using AMI as this sometimes hangs and a full reload is needed to get
the system going again - something I cant do with. Any ideas on how to do
the reload - and know that it is done world be welcome.

Paddy

 

-Original Message-
From: Steve Edwards [mailto:asterisk@sedwards.com] 
Sent: 14 March 2011 19:26
To: pa...@wizaner.com; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Asterisk -rx command not returning data -
Version 1.4.33.1

On Mon, 14 Mar 2011, Paddy Grice wrote:

 I am having trouble running the command
  
 siptest:~# asterisk -rx 'dialplan reload'
  
 I assume the problem is timing but any ideas on how to fix it

I'm just a 1.2 Luddite, but it's been my experience that issuing shell
command lines and parsing the output is unreliable. Kind of hit or miss,
sometimes you get more that you expect, sometimes less.

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


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[asterisk-users] Problems using Background within a macro on V 1.4

2011-02-02 Thread Paddy Grice 
Hi List
 
I have had a look at the various posts on this and seem to be more confused
than ever - but then again that's not hard ;-)
 
I am using Version 1.4.33.1 build from the Debian lenny distros
 
I am trying to implement a simple screening 
 
[macro-screen]
exten = s,1,Background(press1)
exten = s,n,WaitExten(5)
exten = 1,1,NoOp(accepted) ; Dont set a reply so dial connects
exten = i,1,Set(MACRO_RESULT=CONTINUE)
exten = t,1,Set(MACRO_RESULT=CONTINUE)
 
[test]
exten = _1234,1,Dial(SIP/2001,10,M(screen))
exten = _1234,n,playback(sorry,noanswer)
 
This config plays to the caller sorry if I don't answer SIP/2001 - good 
and if I answer SIP/2001 but don't press ANY key (so timeout) the caller
gets the sorry message - again good.
 
The problem I have is that the call gets connected whatever key I press -
not just the 1 key.
 
I have seen various posts about using background within macros e.g.
http://www.voip-info.org/wiki/view/Asterisk+cmd+BackGround
 
but changing the macro to 
 
[macro-screen]
exten = s,1,Background(press1,,screen)

 
or - I am not clear what the name should be 
  
[macro-screen]
exten = s,1,Background(press1,,macro-screen)

 
just produces the same result - If I can get this working it will be part of
a bigger dialplan for which the followme app is not suitable hence the back
to basics
 
can anybody point me in the right direction - pleeese :-)
 
Paddy
 
 
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[asterisk-users] Voicemail - disable * 0 and #

2010-09-02 Thread Paddy Grice 
Hi all 

Been looking to find a way to stop the dtmf keys * 0 and # managing call
flow in the dialplan - I just want VM to stop recording on silence or
hangup.

I know I can trap the exit and loop back around but just want to ignore the
keys totally.

Any suggestions

P




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Re: [asterisk-users] Voicemail - disable * 0 and #

2010-09-02 Thread Paddy Grice 
oops - forgot to say this is voicemail() on Version 1.4.33.1
 
P

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paddy Grice
Sent: 02 September 2010 13:32
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Voicemail - disable * 0 and #



Hi all 

Been looking to find a way to stop the dtmf keys * 0 and # managing call
flow in the dialplan - I just want VM to stop recording on silence or
hangup.

I know I can trap the exit and loop back around but just want to ignore the
keys totally. 

Any suggestions 

P 




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Re: [asterisk-users] Calling Line Identity - any ideas

2010-08-22 Thread Paddy Grice 
Hi All
 
Thanks for the pointers - I now have a working solution using local channels
and for the few occasions this needs to happen, about 300 calls in the
20,000 we handle each day I am very happy.
 
Again thanks for you help
 
Paddy


  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nasir Iqbal
Sent: 22 August 2010 05:23
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Calling Line Identity - any ideas


In simple words , Paddy should go  with my trick,  that is  what i got from
this reply  


Regards



On Sat, Aug 21, 2010 at 5:14 AM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:


Nasir Iqbal na...@ictinnovations.com wrote:

 With all honor and respect you deserve,  Do  I need your permission to
  express my point of view  on community forum ?
 also it would be quiet helpful for us if you understand well
 the requirement of post

*snip*

Nasir,
You don't need my permission to post on a public forum...However,
neither do I, and I took issue with what you said, and found that your
comment about those who are dealing with high load traffic
offensive, since it made the assumption that I was just some new guy
who deals with hobby/small Asterisk systems and doesn't know what he's
talking aboutTherefore, I made it abundantly clear that I wasn't,
and that I definitely took issue with that statement.

However, I will say that yes, I did mis-take something the OP said...

Paddy:
Now, here's idea I came up with (haven't tested yet, too busy writing
a system for an international interpretation company's telecom needs)

First of all, you should have a separate context for outbound calls
made by internal extensions... so, in THAT context have code to set
the CID to what you wish (you can do logic control and if you're
feeling spiffy you can even lookup what CLID to use based on the
extension making the call).

Second, calls that are being passed from the outside world onto should
pass through a different context, performing pretty much the same
function...

Third, both of THOSE contexts should then pass to a third context that
performs the dialout using the multiple targets...


Let me know if that works...I know I can make this do what you want,
but I'm not trying to do all the work, just point you in a direction,
since I get paid to actually do the work ;-)


Cheers all, and remember, some of us have been doing this a while, and
get grumpy... ;-)

   there's still no conceivable reason
  What can be? except performance! (as asterisk has to create one
  additional leg and bridge it) Which is very conceivable to those who
  are dealing with high load traffic.
  And what will be the option, if other outgoing call requires
  different
  custom CLI while using the same trunk?




  New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
 
  asterisk-users mailing list
  To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
  First, the reason is, why use a BAD IDEA when there's perfectly good
  solutions in front of the user There was no mention on this ONE
  call
  going outbound over the trunk needing a different CID...the request
  was as
  follows:
 
  Client needs to call an INTERNAL extension, where the INTERNAL
  CallerID will
  be used, and at the SAME TIME, a call to an EXTERNAL number (which
  would
  necessitate USING THEIR PROVIDER TRUNK), using the EXTERNAL
  CallerID
 
  Now, p-lease tell me how just configuring the damned trunk's
  outbound CID is
  NOT more sensible, efficient, and just friggin' COMMON SENSE TO START
  WITH...over using a Local channel call, which would require slightly
  more
  typing, and using something that I've almost NEVER found a good
  reason to
  use, and if you'd care to search the damn archives, you'll see that
  I was
  pushing upwards of 5k CONCURRENT CALLS back in 2005, WITH 1.4 Trunk
  and the
  RealTime addiiton (which was experimental)...
 
  For the love of whatever you find holy and good and true...don't
  come at me
  like that...I'm really not in the mood anymore...I put 3-4 solid
  years of
  helpjng newbies figure out why shit didn't work, reporting REAL bugs
  and
  issues to thew developers and even assisting with some of the
  fixesI
  feel entitled (yes, I know that's an asshole thing to say) to a little
  common respect
 
 
  Now...anyone for a pint? I'm off to vent some frustration with
  people who
  jump on the WRONG bandwagon and try to take over
 
  Sherwood Mother-F'in' McGowanb...
  Telecommunications and Tattooing
  You konw anyone else who combines those two professions? I'd like to
  buy
  that guy a drink!
 
 
 
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[asterisk-users] Calling Line Identity - any ideas

2010-08-19 Thread Paddy Grice 
Hi list

I have a requirement that I just don't know how to address - I don't think
its strange but can't find any pointers anywhere.

I have a user that wishes to have a multi phone divert. By that I mean 

calls made to his extension say Ext200 can be redirected to a different
extension say Ext400 and also to his home landline.

Doing the dial is fine using Dial(SIP/Ext400SIP/TheWorld/441234567890)

The problem is CLID - 

At the moment internal calls (Ext to Ext) show a CLID EXTxxx and External
Calls show the received CLID.

When the phone is redirected to both Internal and external numbers he wants
the correct CLI displayed on both phones.

So with the redirect operational 

1) a call from the outside world to his DID number will show the received
CLI(ANI) on both devices - this works
BUT 
2) a call from an office extension needs to show EXTxxx on the extension
(Ext400) but show the office telephone number on the landline 

so in pseudo code I want to do something like

Dial ( SIP/Ext400 using CLID EXT123  SIP/TheWorld/441234567890 using CLID
44112233445566 )

Any ideas ?

Paddy

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Re: [asterisk-users] Calling Line Identity - any ideas

2010-08-19 Thread Paddy Grice 
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paddy Grice
Sent: 19 August 2010 08:21
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Calling Line Identity - any ideas



Hi list 

I have a requirement that I just don't know how to address - I don't think
its strange but can't find any pointers anywhere.

I have a user that wishes to have a multi phone divert. By that I mean 

calls made to his extension say Ext200 can be redirected to a different
extension say Ext400 and also to his home landline.

Doing the dial is fine using Dial(SIP/Ext400SIP/TheWorld/441234567890) 

The problem is CLID - 

At the moment internal calls (Ext to Ext) show a CLID EXTxxx and External
Calls show the received CLID. 

When the phone is redirected to both Internal and external numbers he wants
the correct CLI displayed on both phones. 

So with the redirect operational 

1) a call from the outside world to his DID number will show the received
CLI(ANI) on both devices - this works 
BUT 
2) a call from an office extension needs to show EXTxxx on the extension
(Ext400) but show the office telephone number on the landline 

so in pseudo code I want to do something like 

Dial ( SIP/Ext400 using CLID EXT123  SIP/TheWorld/441234567890 using CLID
44112233445566 ) 

Any ideas ? 

Paddy  

 

Forgot to say - I am using Version  1.4.33.1 

Paddy

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Re: [asterisk-users] Calling Line Identity - any ideas

2010-08-19 Thread Paddy Grice 
Hi Sherwood

Maybe my miss-understanding sip.conf I will try and see what happens - but I
don't understand how sip would know which CLID to send to each sip
endpoint - internal or external.

BTW this is all to get return of missed calls working.

I don't know if this makes it any clearer - 

An internal call from Ext123 should send 123 as the CLID to SIP/Ext400 but
should send 442071110123 to SIP/TheWorld but an external call from
44123455667788 should send the received CLID 44123455667788 to both. 

Will check up on sip.conf

Paddy

-Original Message-
From: Sherwood McGowan [mailto:sherwood.mcgo...@gmail.com] 
Sent: 19 August 2010 08:35
To: pa...@wizaner.com; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Calling Line Identity - any ideas

I'd have to say off the top of my head that this should already work as long
as the trunk you're sending calls to the outside world over has the CallerID
setting set AND probably sendrpid=yes...in the sip configuration for both of
those items...past that, I could dig a bit

Cheers,
Sherwood McGowan

On Thu, Aug 19, 2010 at 2:25 AM, Paddy Grice pa...@wizaner.com wrote:
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paddy 
 Grice
 Sent: 19 August 2010 08:21
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Calling Line Identity - any ideas

 Hi list

 I have a requirement that I just don't know how to address - I don't 
 think its strange but can't find any pointers anywhere.

 I have a user that wishes to have a multi phone divert. By that I 
 mean

 calls made to his extension say Ext200 can be redirected to a 
 different extension say Ext400 and also to his home landline.

 Doing the dial is fine using 
 Dial(SIP/Ext400SIP/TheWorld/441234567890)

 The problem is CLID -

 At the moment internal calls (Ext to Ext) show a CLID EXTxxx and 
 External Calls show the received CLID.

 When the phone is redirected to both Internal and external numbers he 
 wants the correct CLI displayed on both phones.

 So with the redirect operational

 1) a call from the outside world to his DID number will show the 
 received
 CLI(ANI) on both devices - this works
 BUT
 2) a call from an office extension needs to show EXTxxx on the 
 extension
 (Ext400) but show the office telephone number on the landline

 so in pseudo code I want to do something like

 Dial ( SIP/Ext400 using CLID EXT123  SIP/TheWorld/441234567890 
 using CLID 44112233445566 )

 Any ideas ?

 Paddy



 Forgot to say - I am using Version  1.4.33.1

 Paddy

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Re: [asterisk-users] Calling Line Identity - any ideas

2010-08-19 Thread Paddy Grice 
Hi Sherwood 

I actually do want dynamic CLID as I tried to make clearer 

 I don't know if this makes it any clearer - 

 An internal call from Ext123 should send 123 as the CLID to SIP/Ext400
but should 
 send 442071110123 to SIP/TheWorld but an external call from
44123455667788 should 
 send the received CLID 44123455667788 to both. 

So over the provider connection the CLID will be different for different
calls. Setting the main office number in sip.conf is fine as a default but
as the code/dialplan needs to set cli for some calls I actually set CLID for
all calls. This setting and onward transmission by provider works fine. 

So what I am trying to do is call 2 different sip endpoints AT THE SAME TIME
presenting different AND VARIABLE CLIs. If Nasir's trick is not recommended
what is the best way to achieve this.

As a newbie to Asterisk advise and best practice gained from user experience
is always welcome.

Paddy



 
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sherwood
McGowan
Sent: 20 August 2010 04:58
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Calling Line Identity - any ideas

On Thu, Aug 19, 2010 at 7:46 PM, Nasir Iqbal na...@ictinnovations.com
wrote:
 Hi,
 there's still no conceivable reason
 What can be? except performance! (as asterisk has to create one 
 additional leg and bridge it) Which is very conceivable to those who 
 are dealing with high load traffic.
 And what will be the option, if other outgoing call requires different 
 custom CLI while using the same trunk?
 Regards
 --
 Nasir Iqbal

 ICT Innovations
 http://www.ictinnovations.com/


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First, the reason is, why use a BAD IDEA when there's perfectly good
solutions in front of the user There was no mention on this ONE call
going outbound over the trunk needing a different CID...the request was as
follows:

Client needs to call an INTERNAL extension, where the INTERNAL CallerID will
be used, and at the SAME TIME, a call to an EXTERNAL number (which would
necessitate USING THEIR PROVIDER TRUNK), using the EXTERNAL CallerID

Now, p-lease tell me how just configuring the damned trunk's outbound CID is
NOT more sensible, efficient, and just friggin' COMMON SENSE TO START
WITH...over using a Local channel call, which would require slightly more
typing, and using something that I've almost NEVER found a good reason to
use, and if you'd care to search the damn archives, you'll see that I was
pushing upwards of 5k CONCURRENT CALLS back in 2005, WITH 1.4 Trunk and the
RealTime addiiton (which was experimental)...

For the love of whatever you find holy and good and true...don't come at me
like that...I'm really not in the mood anymore...I put 3-4 solid years of
helpjng newbies figure out why shit didn't work, reporting REAL bugs and
issues to thew developers and even assisting with some of the fixesI
feel entitled (yes, I know that's an asshole thing to say) to a little
common respect


Now...anyone for a pint? I'm off to vent some frustration with people who
jump on the WRONG bandwagon and try to take over

Sherwood Mother-F'in' McGowanb...
Telecommunications and Tattooing
You konw anyone else who combines those two professions? I'd like to buy
that guy a drink!



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Re: [asterisk-users] Busy Lamp Fields

2010-07-19 Thread Paddy Grice 

Rob Many thanks for the pointer - I was missing limitonpeers=yes in the
general section - Sorry I didn't say version (1.4.33.1) etc forgot with
frustration ;-) 

Paddy


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[asterisk-users] Busy Lamp Fields

2010-07-16 Thread Paddy Grice 
Hi all 

A quick question about busy lamps

I have lamps working 'sort-of'  on my GXP2000 lamps flash with ringing and
go solid red when call gets answered but stay green when a call is made from
the extension.

Setup is Ext 200, 201, 202, each monitor the other two

when 200 calls 202 - the BLF on 200 and 201 flash red - when 202 answers
both 200 and 201 show BLF for 200 as red but neither 201 or 202 show the
calling extension 200 as busy 

Is this normal or have I missed something.

Seems BLF only work on called extensions - is there a way to show busy for
the calling extension?

any help or pointers would be more than welcome

Paddy

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Re: [asterisk-users] VOIP Monitoring tools........

2010-04-25 Thread Paddy Grice 
Hi
 
I use ADSL and SDSL on a lot of multi channel VoIP connections SIP and H323
and no real problems if you size the link correctly - this normally means
limiting no of calls to match available bandwidth.
 
Check out the upstream and downstream data rates and size on the smaller -
normally the upstream. the thing that is not normally explained is the
contention ratio at the local exchange. here in the UK residential broadband
packages normally have a contention ratio of 50:1 so at peak you could be
fighting with 49 other users. Contention ratios for business DSL packages
can vary from 1:1 to typically 20:1.
 
To monitor I use wireshark and look at packet loss on RTP. 
 
My home adsl connection is 7M downstream / 800K upstream with a 10:1
contention.I can normally get 20 x G729 channels without a problem much more
and things get stressed - also I have to watch what other things are going
on - downloads not too much of a problem because of the unbalanced
downstream/upstream speeds but peer to peer stuff is a real killer!
 
Paddy
 
 
 
 
 
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From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mike mosier
Sent: 25 April 2010 02:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] VOIP Monitoring tools


Hey all
 
What VoIP networking monitoring, asterisk monitoring tools would you
recommend? I started working with an IT company that insists on using DSL
with a Sonicwall router. The problem is that the clients are having sound
problems. The owner is convinced that it's the Asterisk box. In the 4 yrs I
have been doing this I have not had this bad a sound problem and it always
came down to a bad setup in the Cisco router. Asterisk just doesn't have
sound problems so I am going to have to convince him that its either the
router or DSL. Has anyone used DSL for SIP traffic? How about Sonicwall
routers?
 
Michael D Mosier

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[asterisk-users] RTP Timeouts not clearing calls

2010-04-19 Thread Paddy Grice 
Hi there

I hope someone can help - I am having a big problem getting calls cleared
from several asterisk systems when RTP timeouts occur.

It appears that asterisk doesn't send a BYE when it decides to terminate a
call because of a RTP Timeout - is this a configuration problem? if so what
need changing? or is this a bug/feature? if so is there a work around?

The setup here is calls come from our Nextone softswitch and connect to
remote Asterisk boxes and this all works fine unless the Asterisk box
decides there is an RTP timeout - it clears the call down as far as the
remote (asterisk) end is concerned but the Nextone never sees the BYE so
holds the channel up until a Max-Duration  timer clears it down. Wireshark
traces on the asterisk box confirm no Bye is sent.

The link between switches is still up and usable as new calls are
established and clear normally. Whilst I know I need to investigate why we
are getting these RTP timeouts the fact that no attempt to clear the call
(from the Asterisk end) is causing me all sort of problems.

I believe the far end version is 1.4

Any help or advice welcome

Thanks - Paddy
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