date:20110520


Hey Eric,

I do have qualify=yes. Am i missing something ?

[seb-exten](!)  ; Template
type=friend
host=dynamic
context=from-sip
qualify=yes
dtmfmode=rfc2833
nat=no
cc_agent_policy=generic
cc_monitor_policy=generic

[7022](seb-exten)
callerid="Rover Conference" <7022>
accountcode="Rover Conference"
mailbox=7022@default

[7023](seb-exten)
callerid="Faire Conference" <7023>
accountcode="Faire Conference"
mailbox=7023@default



> From: ewiel...@nyigc.com
> To: asterisk-users@lists.digium.com
> Date: Fri, 20 May 2011 15:15:45 -0400
> Subject: Re: [asterisk-users] Restart asterisk destroy all registered SIP 
> peers
> 
> 
> 
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> > satish patel
> > Sent: Friday, May 20, 2011 3:10 PM
> > To: asterisk-users
> > Subject: Re: [asterisk-users] Restart asterisk destroy all
> > registered SIP peers
> >
> > Issue is we are running customer support queue and if by
> > chance if i need to restart asterisk then they will not able
> > to get call until phone get register :(  Let me check polycom
> > default timeout and set to min.
> 
> Asterisk should cache the registrations across a restart and reboot.  I 
> belive this feature was added in 1.4.
> 
> You should not need to set a low registration timeout.  If you set it because 
> of NAT issues, setting qualify=yes will keep the translations open.
> 
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Re: [asterisk-users] Faxing with Asterisk 1.8.4 & T.38

2011-05-20 Thread Anthony Messina

On 05/20/2011 01:20 PM, e...@erols.com wrote:
> #1) We would like to use Fax For Asterisk with asterisk 1.8.4 in order to 
> receive faxes via T.38.  Sending faxes is not a requirement.  Does anyone 
> have a working asterisk 1.8.4 configuration and ITSP provider that they can 
> recommend?  We have been trying T.38 DIDs from our current ITSP, but we have 
> been unable to make it work.  I am more than happy to purchase new DIDs from 
> a different provider if they will consistently work and are fairly priced.

I use http://www.ipcomms.net/ with a free inbound DID for faxes.  I
always receive T.38.

I use http://www.gafachi.com/ for outbound T.38.

I have had excellent service from both.

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



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Re: [asterisk-users] Restart asterisk destroy all registered SIP peers

2011-05-20 Thread Eric Wieling



> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> satish patel
> Sent: Friday, May 20, 2011 3:10 PM
> To: asterisk-users
> Subject: Re: [asterisk-users] Restart asterisk destroy all
> registered SIP peers
>
> Issue is we are running customer support queue and if by
> chance if i need to restart asterisk then they will not able
> to get call until phone get register :(  Let me check polycom
> default timeout and set to min.

Asterisk should cache the registrations across a restart and reboot.  I belive 
this feature was added in 1.4.

You should not need to set a low registration timeout.  If you set it because 
of NAT issues, setting qualify=yes will keep the translations open.

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Re: [asterisk-users] Restart asterisk destroy all registered SIP peers

Issue is we are running customer support queue and if by chance if i need to 
restart asterisk then they will not able to get call until phone get register 
:(  Let me check polycom default timeout and set to min.

-S

From: mden...@gmail.com
Date: Fri, 20 May 2011 15:03:35 -0400
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Restart asterisk destroy all registered SIP   
peers

On Fri, May 20, 2011 at 3:00 PM, satish patel  wrote:

We have polycom 501 and i am waiting since last 5 min no registration require 
appear. 

-S

With Polycom 321 you can poke around the menus -- one of them has a countdown 
timer which will show you when the next registration happens.

-M 

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Re: [asterisk-users] Restart asterisk destroy all registered SIP peers

2011-05-20 Thread Mark Deneen

On Fri, May 20, 2011 at 3:00 PM, satish patel  wrote:

>  We have polycom 501 and i am waiting since last 5 min no registration
> require appear.
>
> -S
>
>
With Polycom 321 you can poke around the menus -- one of them has a
countdown timer which will show you when the next registration happens.

-M
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Re: [asterisk-users] Restart asterisk destroy all registered SIP peers

We have polycom 501 and i am waiting since last 5 min no registration require 
appear. 

-S

From: mden...@gmail.com
Date: Fri, 20 May 2011 14:56:20 -0400
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Restart asterisk destroy all registered SIP   
peers

On Fri, May 20, 2011 at 2:10 PM, satish patel  wrote:

Hi Guys!

This is strange issue with 1.8 I have restarted my asterisk and it destroy all 
registered SIP peers now only solution is i manually reboot all phones to get 
them register back. I have never seen issue like this before. Any idea what 
would be the issue ?

Thanks
S
Shouldn't the phones re-register on their own?  Mine do it every few minutes.
-M 

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Re: [asterisk-users] Restart asterisk destroy all registered SIP peers

2011-05-20 Thread Mark Deneen

On Fri, May 20, 2011 at 2:10 PM, satish patel  wrote:

>  Hi Guys!
>
> This is strange issue with 1.8 I have restarted my asterisk and it destroy
> all registered SIP peers now only solution is i manually reboot all phones
> to get them register back. I have never seen issue like this before. Any
> idea what would be the issue ?
>
> Thanks
> S
>

Shouldn't the phones re-register on their own?  Mine do it every few
minutes.

-M
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[asterisk-users] Playback noanswer & SIP

2011-05-20 Thread Jorge Mendoza

Hi,
I would to send a message to an incoming call with no answer. My Asterisk 
server receive the incoming call by a BRI/SIP gateway (Could be a PRI, for 
instance).
I do the command playback with option noanswer, Asterisk send 183 followed by 
RTP and finish with 603. But the BRI gateway do not allow to pass the RTP 
without a 200 OK.
The question is: are there a SIP command to indicate the gateway to allow pass 
the RTP without the 200?
This is an usual case when the Service Provider play a message like "I'm sorry, 
you have dialed a wrong number...". So, I assume that the SIP protocol have 
foreseen the commands to implement this feature, I hope.
Thank You
--
Jorge Mendoza

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[asterisk-users] Faxing with Asterisk 1.8.4 & T.38

2011-05-20 Thread e...@erols.com

Hi -

I am looking for suggestions for ITSPs for faxing with asterisk 1.8.  We are 
based in the US, so would need an ITSP with US DIDs.

#1) We would like to use Fax For Asterisk with asterisk 1.8.4 in order to 
receive faxes via T.38.  Sending faxes is not a requirement.  Does anyone have 
a working asterisk 1.8.4 configuration and ITSP provider that they can 
recommend?  We have been trying T.38 DIDs from our current ITSP, but we have 
been unable to make it work.  I am more than happy to purchase new DIDs from a 
different provider if they will consistently work and are fairly priced.

#2) We would also like to connect an ATA directly to some of our fax machines, 
and use that for both sending and receiving faxes.  We have several SPA2102s 
(and similar models) and have been unable to get them to work with T.38 and our 
current provider.  We have tried connecting the ATAs directly to our ITSP, and 
also with asterisk sitting in the middle doing pass-through. Once again, I am 
willing to purchase new DIDs, and purchase new ATAs if someone can point us 
towards a known working provider & configuration.

Any help or suggestions would be appreciated.  

Thanks!

Eric
e...@erols.com


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[asterisk-users] Restart asterisk destroy all registered SIP peers


Hi Guys!

This is strange issue with 1.8 I have restarted my asterisk and it destroy all 
registered SIP peers now only solution is i manually reboot all phones to get 
them register back. I have never seen issue like this before. Any idea what 
would be the issue ?

Thanks
S
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Re: [asterisk-users] Agent (Invalid) has taken no calls yet

2011-05-20 Thread Carlos Chavez

On Fri, 2011-05-20 at 17:57 +, satish patel wrote:
> 
> I do have agents in agents.conf. I am not using agentlogin apps. I am
> using AddQueueMember 
> 
> agent => 7101,,Agent1
> agent => 7102,,Agent2
> 
> 
> 
> From: cur...@telecomabmex.com
> To: asterisk-users@lists.digium.com
> Date: Fri, 20 May 2011 11:56:23 -0500
> Subject: Re: [asterisk-users] Agent (Invalid) has taken no calls yet
> 
> On Thu, 2011-05-19 at 21:10 +, satish patel wrote:
> > How to get rid on following.. why its Invalid ? 
> > 
> > holler*CLI> queue show queue1
> > queue1 has 0 calls (max unlimited) in 'rrmemory' strategy (0s
> > holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
> >Members:
> >   Agent/7201 (Invalid) has taken no calls yet
> >   Agent/7202 (Invalid) has taken no calls yet
> >No Callers
> > 
> > 
>  
>   Your agents are invalid because they are not pointing to a valid
> device.  Is the agent defined in agents.conf?  When the agent logs in is
> he/she passing the correct extension to agentlogin?
>  
>   Maybe it is time to consider dynamic agents for your queues?  Since
> agentcallbacklogin was deprecated in 1.6 I think static agents are more
> of a bother than they are worth.
>  
If you are using dynamic agents then you do not need agents.conf.  You
also do not need to specify members in queues.conf.  Just use the device
(like SIP/7201) when you add the dynamic queue member.  

-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] Agent (Invalid) has taken no calls yet



I do have agents in agents.conf. I am not using agentlogin apps. I am using 
AddQueueMember 

agent => 7101,,Agent1
agent => 7102,,Agent2



From: cur...@telecomabmex.com
To: asterisk-users@lists.digium.com
Date: Fri, 20 May 2011 11:56:23 -0500
Subject: Re: [asterisk-users] Agent  (Invalid) has taken no calls yet

On Thu, 2011-05-19 at 21:10 +, satish patel wrote:
> How to get rid on following.. why its Invalid ? 
> 
> holler*CLI> queue show queue1
> queue1 has 0 calls (max unlimited) in 'rrmemory' strategy (0s
> holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
>Members:
>   Agent/7201 (Invalid) has taken no calls yet
>   Agent/7202 (Invalid) has taken no calls yet
>No Callers
> 
> 
 
Your agents are invalid because they are not pointing to a valid
device.  Is the agent defined in agents.conf?  When the agent logs in is
he/she passing the correct extension to agentlogin?
 
Maybe it is time to consider dynamic agents for your queues?  Since
agentcallbacklogin was deprecated in 1.6 I think static agents are more
of a bother than they are worth.
 
-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001

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Re: [asterisk-users] Agent (Invalid) has taken no calls yet

2011-05-20 Thread Carlos Chavez

On Thu, 2011-05-19 at 21:10 +, satish patel wrote:
> How to get rid on following.. why its Invalid ? 
> 
> holler*CLI> queue show queue1
> queue1 has 0 calls (max unlimited) in 'rrmemory' strategy (0s
> holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
>Members:
>   Agent/7201 (Invalid) has taken no calls yet
>   Agent/7202 (Invalid) has taken no calls yet
>No Callers
> 
> 

Your agents are invalid because they are not pointing to a valid
device.  Is the agent defined in agents.conf?  When the agent logs in is
he/she passing the correct extension to agentlogin?

Maybe it is time to consider dynamic agents for your queues?  Since
agentcallbacklogin was deprecated in 1.6 I think static agents are more
of a bother than they are worth.

-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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[asterisk-users] Static agent in queue


Hi,

I want to add static agent in queue so how to do that it seem 1.8 has very 
different approach. I have added SIP extension but they are not getting calls.


@queues.conf 

member => SIP/blah   
member => SIP/blah 
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Re: [asterisk-users] click to call with php

2011-05-20 Thread salaheddine elharit

thanks a lot for your advice i really appreciate it :)

2011/5/20 Mark Deneen 

>
>
>  On Fri, May 20, 2011 at 4:38 AM, Ishfaq Malik  wrote:
>
>> If you are going to use call files don't write them directly to
>>
>> /var/spool/asterisk/outgoing/
>>
>> write them in some temp directory and then move them to
>>
>> /var/spool/asterisk/outgoing/
>>
>> Ish
>>
>>
>
> Make sure that your temp file is on the same mounted file system as
> /var/spool/asterisk/outgoing.  If they are on different file systems, mv
> will do a cp and a rm in this situation and you won't get the atomic
> operation you were hoping for.
>
> -M
>
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Re: [asterisk-users] AstManProxy

On Fri, 2011-05-20 at 11:58 -0400, Jose P. Espinal wrote:
> > Is this the same as AJAM?
> > http://www.voip-info.org/wiki/view/Aynchronous+Javascript+Asterisk
> > +Manager+(AJAM)
> 
> Yes, but I would definitively look for information in 
> http://wiki.asterisk.org as voip-info is getting an paleontological feel 
> now.
> 
> If you have any doubts, let me know in order to take some notes about 
> common questions. I'll be writing some docs soon about using AJAM with 
> PHP and Javascript (JQuery).
> 
> Regards,
> 
Thanks Jose, usually all I need is pointing in the right direction :)

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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Re: [asterisk-users] AstManProxy

2011-05-20 Thread Jose P. Espinal


Is this the same as AJAM?
http://www.voip-info.org/wiki/view/Aynchronous+Javascript+Asterisk
+Manager+(AJAM)


Yes, but I would definitively look for information in 
http://wiki.asterisk.org as voip-info is getting an paleontological feel 
now.


If you have any doubts, let me know in order to take some notes about 
common questions. I'll be writing some docs soon about using AJAM with 
PHP and Javascript (JQuery).


Regards,

--
Jose P. Espinal
http://www.eslackware.com
IRC: [OFTC|FreeNode]
Khratos @ #slackware | #asterisk/-doc/-bugs

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Re: [asterisk-users] AstManProxy

On Fri, 2011-05-20 at 11:27 -0400, Jose P. Espinal wrote:
> 
> > Building a web page which uses AJAX to get information from the AMI
> > every 10-30 seconds or so and not wanting to log on and off via AMI that
> > many times.
> 
> You could use HTTP AMI inteface. You would need to login only once, 
> start a session and send requests to Asterisk.
> 
> I've used this approach in PHP, becasue (as good as it is for the web) 
> it's not a very good option to work with generic sockets. I've been able 
> to send an considerable amount of requests per minutes (like 20 - 50) to 
> Asterisk, and worked Ok without problems.
> 
> In your case, one request per 10-30 seconds should be just fine.
> 
> Regards,
> 
Is this the same as AJAM?
http://www.voip-info.org/wiki/view/Aynchronous+Javascript+Asterisk
+Manager+(AJAM)
-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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Re: [asterisk-users] AstManProxy

2011-05-20 Thread Earl Terwilliger

On Friday, May 20, 2011 10:46:05 AM Ishfaq Malik wrote:
> On Fri, 2011-05-20 at 10:58 -0400, Leif Madsen wrote:
> > On 11-05-20 09:37 AM, Ishfaq Malik wrote:
> > > Do many people use this?
> > > Is it reliable and safe?
> > 
> > It may still work, but that code is quite old, and I'm not even sure it's
> > necessary any more.
> > 
> > Leif.
> 
> The reasons I'm considering it are as follows:
> Building a web page which uses AJAX to get information from the AMI
> every 10-30 seconds or so and not wanting to log on and off via AMI that
> many times.
> 
> We'll soon be using multiple asterisk servers so having a single point
> of access would be very useful.
> 
> I'd love for you to elaborate on why it's not necessary any more, is
> there something simple I've overlooked?

You might find this of interest:  http://www.micpc.com/eventmonitor/

It has a python script which does basically the same thing as AstManProxy and 
uses AJAX to get info from the database that it writes into.

It was written because, as you mention, you don't want to have to poll the AMI 
every so often (see the README.TXT)

earl

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Re: [asterisk-users] AstManProxy

2011-05-20 Thread Gordon Henderson


On Fri, 20 May 2011, Leif Madsen wrote:


On 11-05-20 09:37 AM, Ishfaq Malik wrote:

Do many people use this?
Is it reliable and safe?


It may still work, but that code is quite old, and I'm not even sure it's
necessary any more.


I had to set it up once for some CRM plugin (Possibly Sugar) that wanted 
to use it, rather than talk directly to the manager port...


I really couldn't work out why they wanted one more layer of software to 
pass commands through, but there you go.


Gordon

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Re: [asterisk-users] AstManProxy

2011-05-20 Thread Jose P. Espinal





Building a web page which uses AJAX to get information from the AMI
every 10-30 seconds or so and not wanting to log on and off via AMI that
many times.


You could use HTTP AMI inteface. You would need to login only once, 
start a session and send requests to Asterisk.


I've used this approach in PHP, becasue (as good as it is for the web) 
it's not a very good option to work with generic sockets. I've been able 
to send an considerable amount of requests per minutes (like 20 - 50) to 
Asterisk, and worked Ok without problems.


In your case, one request per 10-30 seconds should be just fine.

Regards,

--
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Re: [asterisk-users] ConfBridge - Failed to find a bridge technology to satisfy capabilities

2011-05-20 Thread David Vossel

- Original Message -
> From: "Chris Maciejewski" 
> To: asterisk-users@lists.digium.com
> Sent: Friday, May 20, 2011 8:56:35 AM
> Subject: Re: [asterisk-users] ConfBridge - Failed to find a bridge technology 
> to satisfy capabilities
> > Attach a debug[1] log so we can see what is happening.
> >
> > [1]
> > https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
> 
> debug logs below:
> 
> Asterisk 1.8.4: http://pastebin.com/DFnKgSse
> Asterisk trunk r319661: http://pastebin.com/B19tdbxJ
> 
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Just as a note, the versions of ConfBridge in 1.8 and Trunk are completely 
different.  Trunk will likely give you much better results.  In 1.8 ConfBridge 
is more of just an experimental exercise of the bridging API.

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Re: [asterisk-users] Really, really loud ringers

2011-05-20 Thread Sebastian Arcus

Hi,

On 05/09/2011 09:40 PM, Justin Sherrill wrote:

Anyone have some recommended equipment for alerting people to calls in a noisy
environment?

I have Polycom IP550 phones set up in some really noisy environments - our mine
hoists - and they tend to drown out the ringers. I'm using Clarity WR100s now.
They're analog devices, attached to Linksys PAP2T ATAs as part of a call group
to get a loud (advertised as 95dB) ring out there, but it still could be
louder. Maybe a light-up option would be better.

The old phone system here had some huge loudspeakers that someone had wired
right into the speakers of the old digital phones. I haven't figured out yet
if they need a different voltage, or even if they still work; they were not
responding when I replaced the attached phones.

I've actually used a pc with linphone as a sip client on it, and
amplified speakers attached to the sound card of this computer. I used
this as a dual purpose setup - loud ringer and public announcement
(tannoy in UK) system. I've done a write-up here with details:

http://forum.voxilla.com/asterisk-support-forum/asterisk-public-announcement-system-loud-ringer-bell-49339.html

This way, you can connect your client pc to as loud an amplifier as you
want (or your budget permits) and equally powerful speakers. I think PA
speakers with amplification integrated are quite good for that - but I
settled on a pair of the most powerful active pc speakers I could find -
and they fill the industrial unit where the system is installed nicely.

Sebastian

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Re: [asterisk-users] AstManProxy

On Fri, 2011-05-20 at 10:58 -0400, Leif Madsen wrote:
> On 11-05-20 09:37 AM, Ishfaq Malik wrote:
> > Do many people use this?
> > Is it reliable and safe?
> 
> It may still work, but that code is quite old, and I'm not even sure it's
> necessary any more.
> 
> Leif.
> 
The reasons I'm considering it are as follows:
Building a web page which uses AJAX to get information from the AMI
every 10-30 seconds or so and not wanting to log on and off via AMI that
many times.

We'll soon be using multiple asterisk servers so having a single point
of access would be very useful.

I'd love for you to elaborate on why it's not necessary any more, is
there something simple I've overlooked?
-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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Re: [asterisk-users] SIP Diversion RDNIS - how to get reason parameter?

2011-05-20 Thread Leif Madsen

On 11-05-20 10:39 AM, Benoit Panizzon wrote:
> After reading a bit more in chan_sip.c (I'm not a coder) I fear the result is 
> just put in a temporary variable __SIPDIVERSIONREASON but not in a variable 
> useable in the dialplan.

You could double check by using DumpChan() to see what channel variables are
available for you throughout the dialplan flow.

Also check the CHANNEL() and SIP*() functions to see if there is anything there
that may be of use.

Leif.

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Re: [asterisk-users] AstManProxy

2011-05-20 Thread Leif Madsen

On 11-05-20 09:37 AM, Ishfaq Malik wrote:
> Do many people use this?
> Is it reliable and safe?

It may still work, but that code is quite old, and I'm not even sure it's
necessary any more.

Leif.

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Re: [asterisk-users] ConfBridge - Failed to find a bridge technology to satisfy capabilities

2011-05-20 Thread Terry Wilson

> debug logs below:
> 
> Asterisk 1.8.4: http://pastebin.com/DFnKgSse
> Asterisk trunk r319661: http://pastebin.com/B19tdbxJ

These show that a proper bridging tech module cannot be found to run 
ConfBridge. The debug message showing that a capability for ulaw couldn't be 
found was a buggy debug message which has now been fixed (it isn't a codec 
capability that can't be found, but a bridge capability). You need to make sure 
the bridge_softmix.so is loaded. I would load the other bridge_*.so modules too 
just for fun.
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Re: [asterisk-users] ConfBridge - Failed to find a bridge technology to satisfy capabilities

2011-05-20 Thread Michael L. Young

- Original Message -
> From: "Chris Maciejewski" 
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Sent: Thursday, May 19, 2011 9:39:57 AM
> Subject: [asterisk-users] ConfBridge - Failed to find a bridge technology to  
> satisfy capabilities
> 
> Hi,
> 
> I am trying to use ConfBridge application, but it throws "Failed to
> find a bridge technology to satisfy capabilities 0x4 (ulaw)" error.
> Please see console output below.
> 
> -- Executing [501@services:9] ConfBridge("SIP/OpenSER-0005",
> "1001") in new stack
> [May 19 13:36:05] DEBUG[7452]: app_confbridge.c:404
> join_conference_bridge: Trying to find conference bridge '1001'
> [May 19 13:36:05] DEBUG[7452]: bridging.c:475 ast_bridge_new: Failed
> to find a bridge technology to satisfy capabilities 0x4 (ulaw)
> [May 19 13:36:05] DEBUG[7452]: app_confbridge.c:368
> destroy_conference_bridge: Destroying conference bridge '1001'
> [May 19 13:36:05] ERROR[7452]: app_confbridge.c:435
> join_conference_bridge: Conference bridge '1001' could not be
> created.
> 

I wonder if this recent commit to the 1.8 branch would help fix this issue at 
least with 1.8.

  Author: twilson
  Date: Thu May 19 18:28:13 2011
  New Revision: 319920

  URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=319920
  Log:
  Revert part of a change to the bridging API code

  The capabilities used in the bridging API are very different than the
  ones used for formats. When the conversion was made expanding the bit
  width of codecs, the bridging code was accidentally accosted in ways
  that it didn't deserve.

  Modified:
  branches/1.8/include/asterisk/bridging.h
  branches/1.8/include/asterisk/bridging_technology.h
  branches/1.8/main/bridging.c

As far as why svn trunk is stating that it cannot create the bridge, the debug 
message is not very informative as to why it couldn't create the conference 
bridge from what I could see briefly looking at the debug logs you posted in 
another message.

Michael
(elguero)

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Re: [asterisk-users] Hints custom:abcdef

2011-05-20 Thread Jonas Kellens

In other words : is it correct to say that hints need to be unique, even 
if they are defined in different contexts ?





On 05/20/2011 12:07 PM, Jonas Kellens wrote:

Hello list,

I want certain devices to monitor certain extensions/SIPaccounts and 
other devices to monitor other extensions/SIPaccounts.


Therefore I do the following :

[from-TEST1]

include => test1-blf

[from-TEST2]

include => test2-blf

[test1-blf]
exten => 10,hint,SIP/testcorp1
exten => 20,hint,SIP/testcorp2

[test2-blf]
exten => 10,hint,SIP/testcorp110
exten => 20,hint,SIP/testcorp120


SIPaccounts with a context definition of "from-TEST1" can not monitor 
the extensions of test2-blf.
SIPaccounts with a context definition of "from-TEST2" can not monitor 
the extensions of test1-blf.



This works great. But now I have a problem with custom hints.

When I do the following :

[test1-blf]
exten => 10,hint,SIP/testcorp1
exten => 20,hint,SIP/testcorp2
exten => 80,hint,custom:light1

I see that SIPaccounts which enter the dialplan in context 
[from-TEST2] also see the state (Green/Red) of hints defined in test1-blf.


So how can I make a difference between custom hints in one context and 
custom hints in another context ??


Is there something like : 
Set(DEVSTATE(*Custom:light1@test1-blf*)=INUSE)* *  ??


Or can there be only one custom:light1 label ??



Kind regards,
Jonas.


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Re: [asterisk-users] SIP Diversion RDNIS - how to get reason parameter?

2011-05-20 Thread Benoit Panizzon

Hi Olivier

> Are those PBXs directly connected to each other through a SIP trunk ?

Yes, and the reason is definitely transmitted in the SIP header and also 
parsed by asterisk and displayed in debug output.

After reading a bit more in chan_sip.c (I'm not a coder) I fear the result is 
just put in a temporary variable __SIPDIVERSIONREASON but not in a variable 
useable in the dialplan.

Kind regards

Benoit Panizzon
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Re: [asterisk-users] Agent (Invalid) has taken no calls yet


Great! Satish,

I am middle of migration 1.2 queue in 1.8 thats why i encounter there. if i add 
SIP/XXX then my queue working fine. Also i don't understand relation between 
agents.conf and member => at queues.conf 

let me read that URL and see what i can find there.

-S 

Date: Fri, 20 May 2011 09:58:59 +0530
From: satish4aster...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Agent (Invalid) has taken no calls yet

If you go for 1.8,Don't read from 
http://www.asteriskguru.com/tutorials/queues.html. It is bit backdated 
information. Rather I would suggest you to check 


http://ofps.oreilly.com/titles/9780596517342/asterisk-ACD.html.

Queue members are considered INVALID, if their device status is Invalid. This 
is somewhat an error condition.SIP channels are the only type that provide true 
device state information.

I also suggest you to read 'The agents.conf File' section from given link for 
more information.

[SATISH]

On Fri, May 20, 2011 at 2:40 AM, satish patel  wrote:






How to get rid on following.. why its Invalid ? 

holler*CLI> queue show queue1
queue1 has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 0s 
talktime), W:0, C:0, A:0, SL:0.0% within 0s

   Members:
  Agent/7201 (Invalid) has taken no calls yet
  Agent/7202 (Invalid) has taken no calls yet
   No Callers


  

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Re: [asterisk-users] ConfBridge - Failed to find a bridge technology to satisfy capabilities

2011-05-20 Thread Chris Maciejewski

> Attach a debug[1] log so we can see what is happening.
>
> [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

debug logs below:

Asterisk 1.8.4: http://pastebin.com/DFnKgSse
Asterisk trunk r319661: http://pastebin.com/B19tdbxJ

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Re: [asterisk-users] click to call with php

2011-05-20 Thread Mark Deneen

On Fri, May 20, 2011 at 4:38 AM, Ishfaq Malik  wrote:

> If you are going to use call files don't write them directly to
>
> /var/spool/asterisk/outgoing/
>
> write them in some temp directory and then move them to
>
> /var/spool/asterisk/outgoing/
>
> Ish
>
>

Make sure that your temp file is on the same mounted file system as
/var/spool/asterisk/outgoing.  If they are on different file systems, mv
will do a cp and a rm in this situation and you won't get the atomic
operation you were hoping for.

-M
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Re: [asterisk-users] SIP Diversion RDNIS - how to get reason parameter?

2011-05-20 Thread Olivier

2011/5/20 Benoit Panizzon 

> Hi out there
>
> To play the correct announcement in app_voicemail I whould be able to read
> the
> SIP Diversion Reason which ist sent by another PBX:
>
> Are those PBXs directly connected to each other through a SIP trunk ?
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[asterisk-users] SIP Diversion RDNIS - how to get reason parameter?

2011-05-20 Thread Benoit Panizzon

Hi out there

To play the correct announcement in app_voicemail I whould be able to read the 
SIP Diversion Reason which ist sent by another PBX:

Invite contains:

Diversion: ;reason=no-
answer;privacy=off;counter=1

Asterisk Logs:

RDNIS for this call is is +41315995003 (reason no-answer;privacy=off;counter=1)

>From what I see in the source of chan_sip the variable ${SIPDIVERSIONREASON} 
should be set, but it is empty...
Also ${PRIDIVERSIONREASON} is empty...

I'm using: Asterisk 1.6.2.5-0ubuntu1.3

Any hints?

-Benoit-

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Re: [asterisk-users] *8 pickup and CLI presentation

2011-05-20 Thread Olivier

2011/5/20 Ishfaq Malik 

> Hi
>
> When we use the *8 feature to pick up a call on another extension, the
> phone will only display *8 and *8 is what is stored in the phones
> memory. Is there anything we can do so that when we use *8 the incoming
> caller's CLI will be presented on the screen of the phone and in the
> phones memory?
>
> We are using Snom phones but I'm sure this is an asterisk rather than
> phone issue...
>
> Thanks is advance
>
> Ish
> --
> Ishfaq Malik
> Software Developer
> PackNet Ltd
>
> Office:   0161 660 3062
>
>
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>

Google for "P-asserted-id asterisk".
I've never tried it myself so I can't be more helpful.

Cheers
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[asterisk-users] AstManProxy

Hi

Do many people use this?
Is it reliable and safe?

Tanks

Ish
-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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Re: [asterisk-users] click to call with php

2011-05-20 Thread salaheddine elharit

thanks a lot for your help and support

2011/5/20 A J Stiles 

> On Friday 20 May 2011, salaheddine elharit wrote:
> > Ok thank you so much for all advice
>
> This might help you a bit, too:
>
>  $spool = "/var/spool/asterisk/outgoing/"; # outgoing callfile folder
>
> $filename = "asterisk-" . date("U") . "-" . $_SERVER["REMOTE_PORT"] .
> ".call";
> # this should end up being fairly unique.  (if logging to a database with
> an
> # auto increment field, you can also use mysql_insert_id() as a unique
> # reference)
>
> $src = ... ; # source extension
> # you can determine this based on $_SERVER["REMOTE_ADDR"], which is the
> # IP address of the requesting client, by looking up in an array or a
> database
> $ctxt = ... ; # context
> $dest = ... ; # destination number
> # you probably want to get this from $_REQUEST
>
> $callfile = "Channel: SIP/$src\nContext: $ctxt\nExtension: $dest\nPriority:
> 1\nCallerId: $src\n"; # line break added by email, not used in real life!
> if ($fh = fopen("/tmp/$filename", "w")) {
>fwrite($fh, $callfile);
>fclose($fh);
>system("mv /tmp/$filename $spool");
> }
> else {
>die("Call file creation failed");
> };
> ?>
>
> --
>  AJS
>
> Answers come *after* questions.
>
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Re: [asterisk-users] click to call with php

2011-05-20 Thread A J Stiles

On Friday 20 May 2011, salaheddine elharit wrote:
> Ok thank you so much for all advice

This might help you a bit, too:



-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] Asterisk-cpu utilization > 60 %

I think I managed to solve this issue
The problem lay in the VirtualBox setting for the VM.
I will post the exact setting tomorrow which should help others.
Sorry for being a trouble to others :(

Best regards & have a great weekend.
Sans







On Fri, May 20, 2011 at 3:03 PM, RSCL Mumbai  wrote:

> This seems to be an interesting post:
> http://forums.virtualbox.org/viewtopic.php?t=12903
>
> As per OP's message, CONFIG_HG is indeed 1000
>
> [root@e1 ~]# grep CONFIG_HZ /boot/config-2.6.18-194.3.1.el5
> # CONFIG_HZ_100 is not set
> # CONFIG_HZ_250 is not set
> CONFIG_HZ_1000=y
> CONFIG_HZ=1000
> [root@e1 ~]#
>
>
> Not sure if I need to recompile the kernel.
> I will not be able to do this myself anyways... not competent yet.
>
> Sans
>
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[asterisk-users] Hints custom:abcdef

2011-05-20 Thread Jonas Kellens


Hello list,

I want certain devices to monitor certain extensions/SIPaccounts and 
other devices to monitor other extensions/SIPaccounts.


Therefore I do the following :

[from-TEST1]

include => test1-blf

[from-TEST2]

include => test2-blf

[test1-blf]
exten => 10,hint,SIP/testcorp1
exten => 20,hint,SIP/testcorp2

[test2-blf]
exten => 10,hint,SIP/testcorp110
exten => 20,hint,SIP/testcorp120


SIPaccounts with a context definition of "from-TEST1" can not monitor 
the extensions of test2-blf.
SIPaccounts with a context definition of "from-TEST2" can not monitor 
the extensions of test1-blf.



This works great. But now I have a problem with custom hints.

When I do the following :

[test1-blf]
exten => 10,hint,SIP/testcorp1
exten => 20,hint,SIP/testcorp2
exten => 80,hint,custom:light1

I see that SIPaccounts which enter the dialplan in context [from-TEST2] 
also see the state (Green/Red) of hints defined in test1-blf.


So how can I make a difference between custom hints in one context and 
custom hints in another context ??


Is there something like : 
Set(DEVSTATE(*Custom:light1@test1-blf*)=INUSE)* *  ??


Or can there be only one custom:light1 label ??



Kind regards,
Jonas.
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[asterisk-users] *8 pickup and CLI presentation

Hi

When we use the *8 feature to pick up a call on another extension, the
phone will only display *8 and *8 is what is stored in the phones
memory. Is there anything we can do so that when we use *8 the incoming
caller's CLI will be presented on the screen of the phone and in the
phones memory?

We are using Snom phones but I'm sure this is an asterisk rather than
phone issue...

Thanks is advance

Ish
-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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[asterisk-users] VUC: Sangoma NetBorder 4.0 and an Androis SIP client from Media5 Corp

2011-05-20 Thread randulo

Today, sessions 320-321 of the VoIP Users Conference will take place
at the usual time, 12 Noon Eastern [ http://vuc.me/next for local
times ]

We'll be talking to Sangoma's Frederic Dickey about NetBorder 4.0. You
can download or watch his accompanying slide presentation here:
http://vuc.li/Sangoma-2011

Next, Pascal Doré, mVoIP chief over at Media5 Corporation will talk
about the upcoming Android version of their SIP client Media5fone.

Info and all the various ways to connect are here: http://vuc.me

SIP via g722: 200...@login.zipdx.com (will accept g711, too)
IRC #vuc on Freenode.net or use the web : http://vuc.me/irc

Future Topics : http://vuc.me/future

See you there

:r

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Re: [asterisk-users] click to call with php

2011-05-20 Thread salaheddine elharit

Ok thank you so much for all advice

2011/5/20 A J Stiles 

> On Friday 20 May 2011, Dovid Bender wrote:
> > I had issue with call files. They would lock up the system (this was 5
> > years ago so maybe things have changed.)
>
> Whenever you open a file for writing, a link is created in the containing
> folder's directory  (which says where on the disk the file is located)
> pretty much straight away -- so other processes can see the file.  And
> files
> are written to disk, not one character at a time, but in blocks whose size
> depends on the filesystem, one full block at a time.  The last block may
> well
> be incomplete, and so contain junk after the file proper; but the directory
> entry gives the actual file size, so the junk can be ignored.#
>
> This creates a race condition:  Asterisk may try to parse a call file which
> is
> still in an incomplete state  (empty or just the first block of several),
> and get its knickers in a twist.
>
> The *proper* way to avoid this situation, is to create the call file in a
> temporary location first; then `mv` it to the /var/spool/asterisk/outgoing/
> folder.  Moving a file within a filesystem just entails putting a new link
> in
> the destination folder's directory, and removing the one from the old
> directory.  Moving a file across filesystems entails a copy operation; but
> either way, the important thing is that *the link to the destination file
> won't be placed in the folder's directory until the data is actually
> there*.
>
> The *bodgy* way to avoid this situation, is to make sure the file is
> smaller
> than one logical block on the filesystem where .../outgoing/ resides; turn
> off buffer autoflushing in the scripting language; and cross your fingers
> that the file will already be complete in the cache when the directory is
> updated.  And even if it works on your system today, you might find that an
> upgrade to Asterisk, your scripting language, whatever invoked the script,
> the filesystem driver in the kernel, or even a change in RAM or disk usage
> on
> your server, breaks it tomorrow.
>
> --
> AJS
>
> Answers come *after* questions.
>
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[asterisk-users] first dtmf is not detected

2011-05-20 Thread Szasz Szabolcs

Hi all,

I am using asterisk 1.4.25.1. when I am sending dtmfs the first digit is not
detected.

Do you know a workaround for this?

Besst regards,

Szabolcs Szasz
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Re: [asterisk-users] Asterisk-cpu utilization > 60 %

This seems to be an interesting post:
http://forums.virtualbox.org/viewtopic.php?t=12903

As per OP's message, CONFIG_HG is indeed 1000

[root@e1 ~]# grep CONFIG_HZ /boot/config-2.6.18-194.3.1.el5
# CONFIG_HZ_100 is not set
# CONFIG_HZ_250 is not set
CONFIG_HZ_1000=y
CONFIG_HZ=1000
[root@e1 ~]#


Not sure if I need to recompile the kernel.
I will not be able to do this myself anyways... not competent yet.

Sans
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Re: [asterisk-users] Asterisk-cpu utilization > 60 %

logger.conf is only set for:
full => notice,warning,error,debug
I have now removed "debug".




On Fri, May 20, 2011 at 2:57 PM, Thorsten Göllner  wrote:

>  Maybe IO-Activity caused by intensive logging. Take a look at your
> Log-Files. Maybe one or more log files a growing "rapidly"?
>
> Am 20.05.2011 11:24, schrieb RSCL Mumbai:
>
> CPU utilization is constantly above 24% without any call activity..
>
> *top - 05:53:09 up  1:28,  2 users,  load average: 0.18, 0.27, 0.29
> Tasks:  79 total,   1 running,  78 sleeping,   0 stopped,   0 zombie
> Cpu(s):  9.7%us,  2.3%sy,  0.0%ni, 87.8%id,  0.0%wa,  0.2%hi,  0.0%si,
> 0.0%st
> Mem:   1026824k total,   311300k used,   715524k free,19644k buffers
> Swap:  2064376k total,0k used,  2064376k free,   115668k cached
>
>   PID USER  PR  NI  VIRT  RES  SHR S %CPU %MEMTIME+  COMMAND
>  2154 asterisk  15   0  765m  27m  10m S 24.0  2.7   6:11.13 asterisk
> 1 root  15   0 10348  692  580 S  0.0  0.1   0:03.59 init
> 2 root  RT  -5 000 S  0.0  0.0   0:02.13 migration/0
> 3 root  34  19 000 S  0.0  0.0   0:00.03 ksoftirqd/0
> 4 root  RT  -5 000 S  0.0  0.0   0:08.06 watchdog/0
> 5 root  RT  -5 000 S  0.0  0.0   0:00.09 migration/1*
>
>
>
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Re: [asterisk-users] Asterisk-cpu utilization > 60 %

2011-05-20 Thread Thorsten Göllner



  
  
Maybe IO-Activity caused by intensive logging. Take a look at your
Log-Files. Maybe one or more log files a growing "rapidly"?

Am 20.05.2011 11:24, schrieb RSCL Mumbai:
CPU utilization is constantly above 24% without any
  call activity..
  
  top -
  05:53:09 up  1:28,  2 users,  load average: 0.18, 0.27, 0.29
Tasks:  79
  total,   1 running,  78 sleeping,   0 stopped,   0 zombie
Cpu(s): 
  9.7%us,  2.3%sy,  0.0%ni, 87.8%id,  0.0%wa,  0.2%hi,  0.0%si, 
  0.0%st
Mem:  
  1026824k total,   311300k used,   715524k free,    19644k
  buffers
Swap: 
  2064376k total,    0k used,  2064376k free,   115668k
  cached

  PID
  USER  PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+  COMMAND
 2154
  asterisk  15   0  765m  27m  10m S 24.0  2.7   6:11.13
  asterisk
    1
  root  15   0 10348  692  580 S  0.0  0.1   0:03.59 init
    2
  root  RT  -5 0    0    0 S  0.0  0.0   0:02.13
  migration/0
    3
  root  34  19 0    0    0 S  0.0  0.0   0:00.03
  ksoftirqd/0
    4
  root  RT  -5 0    0    0 S  0.0  0.0   0:08.06
  watchdog/0
    5
  root  RT  -5 0    0    0 S  0.0  0.0   0:00.09
  migration/1


  


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Re: [asterisk-users] Asterisk-cpu utilization > 60 %

CPU utilization is constantly above 24% without any call activity..

*top - 05:53:09 up  1:28,  2 users,  load average: 0.18, 0.27, 0.29
Tasks:  79 total,   1 running,  78 sleeping,   0 stopped,   0 zombie
Cpu(s):  9.7%us,  2.3%sy,  0.0%ni, 87.8%id,  0.0%wa,  0.2%hi,  0.0%si,
0.0%st
Mem:   1026824k total,   311300k used,   715524k free,19644k buffers
Swap:  2064376k total,0k used,  2064376k free,   115668k cached

  PID USER  PR  NI  VIRT  RES  SHR S %CPU %MEMTIME+  COMMAND
 2154 asterisk  15   0  765m  27m  10m S 24.0  2.7   6:11.13 asterisk
1 root  15   0 10348  692  580 S  0.0  0.1   0:03.59 init
2 root  RT  -5 000 S  0.0  0.0   0:02.13 migration/0
3 root  34  19 000 S  0.0  0.0   0:00.03 ksoftirqd/0
4 root  RT  -5 000 S  0.0  0.0   0:08.06 watchdog/0
5 root  RT  -5 000 S  0.0  0.0   0:00.09 migration/1

*
Pls lister, help me, this is driving me crazy...

Thx
Sans
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Re: [asterisk-users] click to call with php

2011-05-20 Thread A J Stiles

On Friday 20 May 2011, Dovid Bender wrote:
> I had issue with call files. They would lock up the system (this was 5
> years ago so maybe things have changed.)

Whenever you open a file for writing, a link is created in the containing 
folder's directory  (which says where on the disk the file is located)  
pretty much straight away -- so other processes can see the file.  And files 
are written to disk, not one character at a time, but in blocks whose size 
depends on the filesystem, one full block at a time.  The last block may well 
be incomplete, and so contain junk after the file proper; but the directory 
entry gives the actual file size, so the junk can be ignored.#

This creates a race condition:  Asterisk may try to parse a call file which is 
still in an incomplete state  (empty or just the first block of several),  
and get its knickers in a twist.

The *proper* way to avoid this situation, is to create the call file in a 
temporary location first; then `mv` it to the /var/spool/asterisk/outgoing/ 
folder.  Moving a file within a filesystem just entails putting a new link in 
the destination folder's directory, and removing the one from the old 
directory.  Moving a file across filesystems entails a copy operation; but 
either way, the important thing is that *the link to the destination file 
won't be placed in the folder's directory until the data is actually there*.

The *bodgy* way to avoid this situation, is to make sure the file is smaller 
than one logical block on the filesystem where .../outgoing/ resides; turn 
off buffer autoflushing in the scripting language; and cross your fingers 
that the file will already be complete in the cache when the directory is 
updated.  And even if it works on your system today, you might find that an 
upgrade to Asterisk, your scripting language, whatever invoked the script, 
the filesystem driver in the kernel, or even a change in RAM or disk usage on 
your server, breaks it tomorrow.

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] [Fwd: FW: realtime mysql - p4]

2011-05-20 Thread Andrew Thomas

Post your cdr_mysql.conf and res_mysql.conf and we'll take it from
there.

Don't forget to remove any 'private' info first (like passwords).

Cheers
 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans
Witvliet
Sent: 19 May 2011 23:14
To: asterisk-users@lists.digium.com
Cc: j.witvl...@mindef.nl
Subject: [asterisk-users] [Fwd: FW: realtime mysql - p4]

Ok, i tried the suggestion:
Instead of:
sippuser => resource, database_name, table_name sippeer  => resource,
database_name, table_name
 
I put in:
sippuser => resource, context, table_name sippeer  => resource, context,
table_name

Unfortunately, with the same results.
btw i tried both "general" as "default"

Besids the commands i tried below, isn't there any other way to see
what's going on?

Perhaps it is totally unrelated, but if i perform a mysql-login on the
prompt, i first have to select the database manualy, ie it isn't
selected by default for the created mysqluser [in this case: voipadmin]

Other wild idea, is there a minimum number of fields that haved to be
filled?

And why is asterisk complaining about not being able to find the
databse, when trying to fill it from the asterisk-CLI?
My database _is_ named "asterisk"..
> kc3054*CLI>  realtime update sipusers set SET port = 4343 WHERE name =
> 0277611 Failed to update. Check the debug log for possible SQL 
> related entries.
> Command 'realtime update sipusers set SET port = 4343 WHERE name = 
> 0277611' failed.
> [May 18 18:47:16] WARNING[16718]: res_config_mysql.c:559 update_mysql:
> MySQL RealTime: Invalid database specified: 'asterisk' (check
res_mysql.conf)

I mean, is that silly or what?


> 
> 
> # grep mysql extconfig.conf |grep sip
> ;sipusers => mysql,asterisk,sip_devices ;sippeers => 
> mysql,asterisk,sip_devices ;sipusers => mysql,general,sip_devices 
> ;sippeers => mysql,general,sip_devices sipusers => 
> mysql,default,sip_devices sippeers => mysql,default,sip_devices
> 
> 
> kc3054*CLI> module show like mysql
> Module Description
Use Count 
> cdr_mysql.so   MySQL CDR Backend
0 
> res_config_mysql.soMySQL RealTime Configuration Driver
0 
> app_mysql.so   Simple Mysql Interface
0 
> 3 modules loaded
> kc3054*CLI>
> kc3054*CLI> sip show users
> Username   Secret   Accountcode
Def.Context  ACL  ForcerPort
> j.witvliet geheimdefault
No   Yes   
> 027761125b06d3a0b5ef73   default
No   Yes   
> kc3054*CLI>
> kc3054*CLI> sip show peers
> Name/username  HostDyn
Forcerport ACL Port Status Realtime
> 0277611(Unspecified)D
N  0Unmonitored 
> j.witvliet (Unspecified)D
N  0Unmonitored 
> 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 
> offline] kc3054*CLI> kc3054*CLI>
> 
> kc3054*CLI>
> kc3054*CLI> realtime mysql cache
> kc3054*CLI> realtime mysql status
> general connected to asterisk@127.0.0.1, port 3306 with username
voipadmin for 18 seconds.
> kc3054*CLI>
> 


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Re: [asterisk-users] click to call with php