[asterisk-users] PHP can't insert - Can someone please help

[bruce bruce]

Hi Guys,

I am making another module for Voicemail. I have three fields in a POST form
that have to be connected together to make it a single 10 digit number but
there is something wrong in my syntax probably.


$npaa = "('$_POST[anpa]')";
$nxxa = "('$_POST[anxx]')";
$blocka = "('$_POST[ablock]')";

*$grplist = $npaa.$nxxa.$blocka;*

$sql="INSERT INTO findmefollow(grpnum, strategy, grptime, grppre, grplist,
annmsg_id, postdest, dring, needsconf, remotealert_id, toolate_id, ringing,
pre_ring)
VALUES 
('$_POST[grpnum]','ringall','$_POST[grptime]','$_POST[grppre]',$grplist,'0','$_POST[postdest]','','','0','0','Ring','$_POST[pre_ring]')";


It seems that $grplist is the problem. Can someone please point what is
wrong?

Error:
Error: You have an error in your SQL syntax; check the manual that
corresponds to your MySQL server version for the right syntax to use near
'('333')(''),'0','ext-local,vmb2000,1','','','0','0','Ring','0')' at
line 3

Thanks,
Bruce
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Re: [asterisk-users] General network question regarding SIP and IAX2

[bruce bruce]

I guess it has to be on the Trunk and one of the either user or peer and the
opposing party shouldn't have it as no.

But, to full proof urself, put it on the trunk and both users. Basically put
it anywhere that takes it.

http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite

<http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite>-Bruce

On Fri, Jul 9, 2010 at 2:40 PM,  wrote:

> Sounds great, thanks for your answer.
> Do i need to set this on the trunk, the friend or on both?
>
>
>
>
>  -----Original Message-
> From: bruce bruce 
> To: Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> Sent: Fri, Jul 9, 2010 8:13 pm
> Subject: Re: [asterisk-users] General network question regarding SIP and
> IAX2
>
>  The variable is *canreinvite.*
> *Please check on voipinfo. If canreinvite is enabled then only SIP
> signaling is passed through Asterisk and the media is not passed through
> Asterisk resulting in less bandwidth usage and probably less jitter buffer,
> etcif you are two phones are closer to each other than a round trip to
> Asterisk server.*
> *
> *
> *On the flip side, you can't record these calls because no media is sent
> through Asterisk.*
> *
> *
> *-Bruce
> *
> On Fri, Jul 9, 2010 at 1:48 PM,  wrote:
>
>> Hi all,
>>
>> i have a beginners question. How are SIP calls and IAX2 calls processed by
>> Asterisk over the network?
>> What i mean is, is there a permanent connection required between the
>> Asterisk Server and the clients or is the Asterisk Server only involved for
>> lets call it the "routing"?
>>
>> From my understanding SIP s used to "find" the "way" to the remote party
>> and the voice is transferred over RTP directly from client to client without
>> permanently involving the Server.
>> IAX seems to do all in one, the "routing" and the transport of the voice.
>>
>> Is that correct?
>>
>> Why i am asking this?
>>
>> Lets say i have one Asterisk running in London and another one in Paris.
>> Both are connected via IAX2 trunk over a WAN connection.
>> User A is registered on the server in London.
>> User B is registered on the server in Paris.
>> Now User A is visiting User B in Paris and both have call with each other.
>> Is the voice data routed from user A to Asterisk in London and then back
>> via IAX2 to the server in Paris and the to user B?
>> Or is there a direct connection between them and no WAN traffic is
>> produced?
>> And is there a difference between using either SIP or IAX as client
>> protocol in that case?
>>
>> I hope i explained well what i meant.
>>
>> Thanks in advance for answers.
>>
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Re: [asterisk-users] General network question regarding SIP and IAX2

[bruce bruce]

The variable is *canreinvite.*
*Please check on voipinfo. If canreinvite is enabled then only SIP signaling
is passed through Asterisk and the media is not passed through Asterisk
resulting in less bandwidth usage and probably less jitter buffer, etcif
you are two phones are closer to each other than a round trip to Asterisk
server.*
*
*
*On the flip side, you can't record these calls because no media is sent
through Asterisk.*
*
*
*-Bruce
*
On Fri, Jul 9, 2010 at 1:48 PM,  wrote:

> Hi all,
>
> i have a beginners question. How are SIP calls and IAX2 calls processed by
> Asterisk over the network?
> What i mean is, is there a permanent connection required between the
> Asterisk Server and the clients or is the Asterisk Server only involved for
> lets call it the "routing"?
>
> From my understanding SIP s used to "find" the "way" to the remote party
> and the voice is transferred over RTP directly from client to client without
> permanently involving the Server.
> IAX seems to do all in one, the "routing" and the transport of the voice.
>
> Is that correct?
>
> Why i am asking this?
>
> Lets say i have one Asterisk running in London and another one in Paris.
> Both are connected via IAX2 trunk over a WAN connection.
> User A is registered on the server in London.
> User B is registered on the server in Paris.
> Now User A is visiting User B in Paris and both have call with each other.
> Is the voice data routed from user A to Asterisk in London and then back
> via IAX2 to the server in Paris and the to user B?
> Or is there a direct connection between them and no WAN traffic is
> produced?
> And is there a difference between using either SIP or IAX as client
> protocol in that case?
>
> I hope i explained well what i meant.
>
> Thanks in advance for answers.
>
> --
> _
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] How to calculate number of speakers needed for PAGING and INTERCOM coverage area?

[bruce bruce]

Thanks fro the input. The area is a 4 square feet. So, you are saying
that if I use four speakers then they would not be as loud as needed?

Thanks again

2010/7/9 Massimo Nuvoli 

> bruce bruce ha scritto:
> > Hi Guys,
> >
> > I am looking to buy a 25 Watt output CyberData VoIP amplifier and to use
> > 2 Bogen sp308a speakers with it for a 40, 000 squar feet area and 21
> > feet height. Is that enough? Is there calculator online I can use to
> > determine the number of speakers needed? I guess these speakers go in
> > chain so I am not sure if the full capacity of the speaker (30 watt)
> > will be used.
>
> Hu interesting... i never checked this kind of product.
>
> CyberData has calculator only for the 8W model, but... every speaker
> they sell is 8w and the calculator say 69 speakers. You can attach 2
> speakers to one amplifier in parallel (they say this also), this is
> the maximum as the amplifier cannot reach 32W (4 speakers), but you
> can try to use 4 speakers (2 in parallel + 2 in parallel) with a
> little less than maximum 8w on each speaker.
>
> I think a reasonable number of speakers may be less than half, but you
> must check wath is in the area, also remember if this is a warehouse
> to place the speakers where a person can be, not goods. :-)
>
> For a so big installation think to use a voip interface and
> professional product with low voltage line speakers, i think is less
> expensive.
>
> Bye.
>
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[asterisk-users] How to calculate number of speakers needed for PAGING and INTERCOM coverage area?

[bruce bruce]

Hi Guys,

I am looking to buy a 25 Watt output CyberData VoIP amplifier and to use 2
Bogen sp308a speakers with it for a 40, 000 squar feet area and 21 feet
height. Is that enough? Is there calculator online I can use to determine
the number of speakers needed? I guess these speakers go in chain so I am
not sure if the full capacity of the speaker (30 watt) will be used.

I appreciate your advice.

Thanks,
Bruce
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Re: [asterisk-users] GURUs - How to monitor all MySQL actions on an Asterisk/FreePBX server?

[bruce bruce]

Putting it in /tmp/ just did the job. Sorry, I posted my older my.cnf file.
I actaully did have the log under mysqld rather than the safe version but it
didn't work. I will put this to privilage problems.

On Thu, Jul 8, 2010 at 9:55 PM, Steve Edwards wrote:

> On Thu, 8 Jul 2010, bruce bruce wrote:
>
> > I have this in /etc/my.cnf:
> >
> > [mysqld]
> > datadir=/var/lib/mysql
> > socket=/var/lib/mysql/mysql.sock
> > user=mysql
> > old_passwords=1
> > log-error=/var/log/mysqld.log
> >
> > [mysqld_safe]
> > log-error=/var/log/mysqld.log
> > pid-file=/var/run/mysqld/mysqld.pid
> > log=/var/log/mysql_query.log
> >
> > But it doesn't log anything to /var/log/mysql_query.log
>
> Move the "log" line to [mysqld] and restart MySQL. Personally, I'd create
> the log file in /tmp/ so there are no permissions issues.
>
> Be aware that using the query log may expose confidential information if
> you have other users and they have read access to the file.
>
> --
> 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] Rx/Tx fine tuning of analogue card to PRI card - Am I right with my theory?

[bruce bruce]

Hi Everyone,

I want to fine tune the Rx and Tx gain on an analogue Sangoma card by
dialing into another server that is running on Sangoma PRI card (both
services on Bell network).

[mwatt1004khz]
exten => s,1,Answer
exten => s,n,PlayTones(1004/1000)
exten => s,n,Wait(300)

If I match the Rx/Tx numbers on both sides by monitoring "ztmonitor X -vv"
am I right with my theory of getting the analogue channel quality as close
to the PRI channel as possible (and in turn get the best quality sound on
analogue lines)?

I can't find a Type 102 Milliwatt for Toronto area to test with but I think
my PRI should be a perfect test system???!!! The milliwatts provided by Bell
talk your number back to you instead of a contineous tone so they can't be
used for this purpose.

Thanks,
Bruce
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Re: [asterisk-users] GURUs - How to monitor all MySQL actions on an Asterisk/FreePBX server?

[bruce bruce]

I have this in /etc/my.cnf:

[mysqld]
datadir=/var/lib/mysql
socket=/var/lib/mysql/mysql.sock
user=mysql
# Default to using old password format for compatibility with mysql 3.x
# clients (those using the mysqlclient10 compatibility package).
old_passwords=1

log-error=/var/log/mysqld.log


[mysqld_safe]
log-error=/var/log/mysqld.log
pid-file=/var/run/mysqld/mysqld.pid
log=/var/log/mysql_query.log


*But it doesn't log anything to /var/log/mysql_query.log*
What else am i missing?

Thanks
On Thu, Jul 8, 2010 at 1:20 AM, Steve Edwards wrote:

> On Thu, 8 Jul 2010, bruce bruce wrote:
>
> > Thanks for the input. Tailing the mysql log file doesn't show me
> > anything even though FreePBX does right to the asterisk table. I think
> > log is more for errors and unexpected shutdowns etcand not queries.
> > In the my.cnf file there is no configuration to higher or lower the
> > verbose to show queries. Any other method of checking queries coming
> > through? Maybe FreePBX uses MySQL direct sockets and its different?
>
> Googling "mysql query log" brings up:
>
> http://dev.mysql.com/doc/refman/5.1/en/query-log.html
>
> as the first hit.
>
> --
> 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] How to know which party hangup() first when using analogue cards? - Dahdi and Asterisk 1.4.x

[bruce bruce]

Hi Everyone,



I am trying to find the issue of dropped calls in the middle of the
conversation. The system is Elastix. Anyway to know which party hangup the
channel in case of Asterisk 1.4 and Sangoma analogue cards? (this is not
PRI)



Thanks,

Bruce
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Re: [asterisk-users] GURUs - How to monitor all MySQL actions on an Asterisk/FreePBX server?

[bruce bruce]

Thanks for the input. Tailing the mysql log file doesn't show me anything
even though FreePBX does right to the asterisk table. I think log is more
for errors and unexpected shutdowns etcand not queries. In the my.cnf
file there is no configuration to higher or lower the verbose to show
queries.

Any other method of checking queries coming through? Maybe FreePBX uses
MySQL direct sockets and its different?

Thanks,
Bruce

On Wed, Jul 7, 2010 at 9:50 PM, Zeeshan Zakaria  wrote:

> On a separate terminal, you can do something like 'tail -f
> /var/log/mysqld.log' or whatever the name of the mysql log file is.
>
> Zeeshan A Zakaria
>
> --
> www.ilovetovoip.com
>
> On 2010-07-07 9:43 PM, "Carlos Chavez"  wrote:
>
>  *On Wed, 7 Jul 2010 19:19:28 -0400, bruce bruce wrote*
>
> > Hi Guys,
> >
> > This is something related and yet un-related to Asterisk. I have a
> FreePBX/Asterisk...
> It is called the mysql query log.  Mysql can create a log of all queries it
> receives.  When using Freepbx Asterisk has no knowledge that Mysql is
> involved (only for CDR) so Asterisk does not send anything configuration
> related to Mysql.
>
> --
> Carlos Chavez
> Director de Tecnología
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Tel: +52-55-91169161 Ext 2001
>
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[asterisk-users] GURUs - How to monitor all MySQL actions on an Asterisk/FreePBX server?

[bruce bruce]

Hi Guys,

This is something related and yet un-related to Asterisk. I have a
FreePBX/Asterisk server running and I want to trace everything that FreePBX
does to MySQL. Is there a verbose CLI to MySQL that I can pull up on
terminal and make configuration change to FreePBX and see it in real-time on
the terminal as to what is added to which MySQL table and where?

Thanks
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Re: [asterisk-users] Y-cords - What are they ?

[bruce bruce]

Thanks for the input guys. My client is looking for Y-cords to train people.
So, set beside them take a call and let them listen on the other call. They
currently use wireless Plantronic headset with Aastra phones. Can you
suggest any specific vendors for Y-cords?

Thanks

On Tue, Jul 6, 2010 at 4:10 PM, Zeeshan Zakaria  wrote:

> We deal with Y-cords all the time for Ethernet and BRIs. They are just
> normal cords, making use of the fact that both Cat5 networks and BRI ports
> don't use all the 8 pins, so why not use extra wires in the cable for
> something useful instead of wasting them. It has nothing to do with the
> performance, and the cables are provided by reputable manufacturers like
> Aculab and Sangoma, because some of their equipment have no choice but to
> use these cables. For example Sangoma's BRI cards use two BRI channels per
> one physical port, so you need one end of the cable with 8 pins and split it
> into two on the other end with 4 pins each. Same is the case on Ehernet
> ports on the Aculab's Groomer II equipment.
>
> Zeeshan A Zakaria
>
> --
> www.ilovetovoip.com
>
> On 2010-07-06 4:00 PM, "Gergo Csibra"  wrote:
>
> Tuesday, July 6, 2010, 8:11:46 PM, bruce wrote:
>
> > Can someone please explain what Y-cords are avail...
>
> I think Y-cords only for PSTN. Or there're Y-cords for twisted pair
> ethetnet too, but that not a good idea.
>
> Usualy VoIP phones includes a mini 2 port switch to use one switch
> port for a phone and a PC.
>
> --
> Best regards,
>  Gergomailto:csi...@gmail.com
>
>
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[asterisk-users] Y-cords - What are they ?

[bruce bruce]

Good Afternoon,

Can someone please explain what Y-cords are available out there and how they
can be used with Aastra or other VoIP phones? Maybe with or WITHOUT
headsets?
Isn't a Y-cord traded for soft Barge in these days?

Thanks,
Bruce
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Re: [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?

[bruce bruce]

Just downloaded PrivateSHELL and it seems to be what everyone is looking for
in Putty. It's much better than putty in terms of not being sluggish and
scrolling is fine. Plus the window and the text doesn't hurt your eyes. It
has One click SFTP as well. So, good bye to WinSCP.

I think I found what I need. I just downloaded it's version 3.0 beta and I
already love it.

Thanks for the suggestion Michael.

-Bruce

On Tue, Jul 6, 2010 at 8:26 AM, Matt Watson  wrote:

>
>
> On Tue, Jun 29, 2010 at 10:39 AM, William Stillwell (Lists) <
> william.stillwell-li...@ablebody.net> wrote:
>
>> I use SecureCRT+FX , and use ansi graphics.
>>
>> Putty is nice w/WinSCP as well.
>>
>>
>>
>
> I'll +1 this - SecureCRT+FX is the first thing I got my employer to buy a
> license of for me when I had to start using Windows on my desktop for other
> reasons instead of a Linux Distro.  I do keep a copy of PuTTY handy on too
> though, the great thing about putty is that it doesn;t require it to be
> installed on a desktop, so you can just keep a copy of the executable on a
> USB flash drive or Windows share that you can then run from any desktop if
> you happen to be at a computer other than your own.
>
> PuTTY works well, but there are some things in it that just drive me
> absolutely crazy, like right-click in the window is an automatic paste... i
> have pasted into a putty window by accident more times than I can count,
> most of the time its from doing a right-click expecting a context menu to
> get a 'copy' action and then i just end up pasting what i actually wanted to
> copy :|
>
> --
> Matt
>
> --
> Matt
>
> --
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Re: [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?

[bruce bruce]

And the 20k+ lines is where it's really hard to handle. The scroll bar is
too small and I was wishing there was an easy page up or page down function
maybe to it rather than using the mouse.

Thanks for the input.

On Tue, Jun 29, 2010 at 11:13 AM, Danny Nicholas  wrote:

> I use PUTTY 0.58 and have Window title and scroll control for 20K+ lines.
> It could use some improvements, but it is more than adequate for "green
> screen" control.  The quality of Putty and many other applications depends
> on how you choose to control it.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roderick A.
> Anderson
> Sent: Tuesday, June 29, 2010 10:08 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] What TERMINAL software do you use for MS
> Windows platform and WHY?
>
> On 06/29/2010 06:53 AM, bruce bruce wrote:
> > Hi Everyone,
> >
> > I am accustomed to PUTTY and it's very nice as in it allows many many
> > SSH profiles to be saved and allows tunneling etcbut it's not very
> > good when it comes to scrolling up and down, colors, text size, and
> > specially it doesn't give a title to the opened instance. Maybe giving
> > the IP address as the title of the window would help a lot if you have
> > many different servers opened at the same time.
>
> I haven't used Putty for several months and that was with a setup I'd
> made several years ago so I can't, off the top of my head, tell you how
> I did it; but I had the remote system's name or IP in the window title
> bar.  It might nave been the name I saved the connection as.
>
> Look in the configuration under Terminal.  Something like a %s and make
> sure the terminal type is either Linux or ANSI.  Again too long ago.
>
>
> Rod
> --
> >
> > Can you please weigh in and tell me what your favorite terminal software
> > is and why?
> >
> > Thanks,
> > Bruce
> >
>
>
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Re: [asterisk-users] Anyway to know when a channel is going to hangup if Dial Timeout option is used?

[bruce bruce]

Thanks for the input Steve.

I don't have access to that part of the dial. I come into the system from a
different context and that channel is already up. It's for me to identify
that channel and then see when it's EPOCH expiring. Even if I had that
channel Unique ID number, can I find out it's remaining EPOCH time?

Thanks

On Sun, Jul 4, 2010 at 12:03 PM, Steve Edwards wrote:

> On Sun, 4 Jul 2010, bruce bruce wrote:
>
>  I have a channel that is dialed with Timeout option. So, there is definite
>> time to it. Only thing is that I don't have control of that channel. I only
>> know that it's using g729 codec and that there is only one channel that is
>> using g729 at any given time. So, my question is:
>>
>> From within the dialplan is there anyway that I can find out at what EPOCH
>> time or normal time is the above mentioned channel going to hangup (since it
>> has a Timeout option)?
>>
>
> When you make the g729 call, add the absolute timeout interval to EPOCH and
> save it in a global variable.
>
> --
> 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] Why does my IAX2 trunk between two office hangup a channel after 30 seconds? Can you share your IAX2 trunking configuration? URGENT HELP much appreciated

[bruce bruce]

Hi guys,

I have two Asterisk servers (with FreePBX) connected together with IAX2
trunking. When I call from server A->B call connects but hangs up after 30
seconds. What could be cause?

Can anyone please share working configuration between two asterisk server in
IAX2 trunking for FreePBX?

Thanks a lot
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Re: [asterisk-users] What are the guts of AsteriskNOW and how it compares to other popular flavors available?

[bruce bruce]

Anything guys?

Thanks

On Mon, Jun 28, 2010 at 10:20 PM, bruce bruce  wrote:

> Hi Everyone,
>
> I want to know a bit about the guts of the current AsterisNOW system. I
> know that FreePBX is embraced as the main GUI but is just an install of
> CentOS 5.4 + (Asterisk/FreePBX from Yum repos)?
>
> - Or is there anymore to this? Maybe some security tools?
> - Or is Asterisk built from the source?
> - Is there some other program installed to facilitate a better PBX system
> than those of the competitor flavours (e.g. Trixbox, piaf, Elastix)? or does
> it generally fall behind those?
> - I would be very happy if you can point me to section of the image file
> which is responsible for post install of CentOS as I think that is the
> approach to get an AsteriskNOW system running with Asterisk and etc...
>
> Thanks,
> Bruce
>
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[asterisk-users] Anyway to know when a channel is going to hangup if Dial Timeout option is used?

[bruce bruce]

Hi Guys,

I have a channel that is dialed with *Timeout* option. So, there is definite
time to it. Only thing is that I don't have control of that channel. I only
know that it's using g729 codec and that there is only one channel that is
using g729 at any given time. So, my question is:

>From within the dialplan is there anyway that I can find out at what EPOCH
time or normal time is the above mentioned channel going to hangup (since it
has a *Timeout* option)?

Thanks,
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Re: [asterisk-users] Problem in establish call from a2billing users.

[bruce bruce]

Yes, you are missing a whole bunch of configurations from creating SIP users
to making sure they show as peers on Asterisk to making sure you use dnid,
etc.You probably might want to search google for some configuration help

On Wed, Jun 30, 2010 at 11:24 AM, gokulakrishnan wrote:

> Hi All,
>
> I installed a2billing with asterisk FreePBX  .  I can able to login and
> make a call with FreePBX but
>
> when i am using the users which is created in a2billing the call was not
> established . I know somewhere i missed
>
> the configuration please any one help me to resolve this issue . Thanks in
> advance.
>
> regards,
>
> gokul.,
>
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Re: [asterisk-users] Anyone can share their config file for Cisco phone please?

[bruce bruce]

Thanks a lot. I will look into it.

On Wed, Jun 30, 2010 at 11:15 AM, Warren Selby wrote:

> On Wed, Jun 30, 2010 at 8:40 AM, bruce bruce  wrote:
>
>> Thanks a lot.
>>
>> -Bruce
>>
>>
>> On Wed, Jun 30, 2010 at 4:55 AM, Emanuele Carbone wrote:
>>
>>> Hi bruce,
>>>
>>> SIPDefault.conf
>>>
>>>
> I think you need one of the newer XML config files for the 7965.  I have an
> example that works with a 7941 on my website (you can find the link my
> signature), I think with a little adaptation you can make it work with a
> 7965.
>
>
> --
> Thanks,
> --Warren Selby
> http://www.selbytech.com
>
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Re: [asterisk-users] Anyone can share their config file for Cisco phone please?

[bruce bruce]

Thanks a lot.

-Bruce

On Wed, Jun 30, 2010 at 4:55 AM, Emanuele Carbone wrote:

> Hi bruce,
>
> SIPDefault.conf
>
> #Image Version
> image_version:P0S3-08-8-00
>
> #Proxy server address
>
>
> # Emergency Proxy info
> proxy_emergency: "192.168.20.4"
> proxy_emergency_port: "5060"
>
> # Backup Proxy info
> proxy_backup: "192.168.20.4"
> proxy_backup_port: "5060"
>
> # NAT/Firewall Traversal
> nat_enable: "0"
> nat_address: ""
> voip_control_port: "5060"
> start_media_port: "16384"
> end_media_port:  "32766"
> nat_received_processing: "0"
>
> telnet_level: "2"
>
> # Time Server  Set time zone to your location
> # Currently on this system the tz is GMT
> sntp_mode: "unicast"
> sntp_server: "192.168.20.4"
> time_zone: "CET"
> dst_offset: "1"
> dst_start_month: "Mar"
> dst_start_day: ""
> dst_start_day_of_week: "Sun"
> dst_start_week_of_month: "4"
> dst_start_time: "2"
> dst_stop_month: "Oct"
> dst_stop_day: ""
> dst_stop_day_of_week: "Sun"
> dst_stop_week_of_month: "4"
> dst_stop_time: "3"
> dst_auto_adjust: "1"
>
> enable_vad : 1
>
> date_format : "D/M/Y"
>
> directory_url: "http://192.168.20.4/xmlservices/phonebook.xml";
>
> logo_url: "http://192.168.20.4/images/logo.bmp";
>
> SIP_MAC_ADDR.conf
>
> proxy1_address: 192.168.20.4
>
> ; Line 1 phone number
> line1_name : 246
>
> ; Line 1 name for authentication with proxy server
> line1_authname : 246
>
> ; Line 1 authentication name password
> line1_password : afjhajshdga
>
> ; Phone Label (Text desired to be displayed in upper right corner)
> phone_label: "XX246"
>
>
> i hope this help you!
>
> regards
>
> 2010/6/30 bruce bruce 
>
>> I have an *ipphone 7965G* which has to be connected to Asterisk. It has
>> been flashed with SIP firmware but the config file doesn't seem to work
>> maybe I am missing something in it.
>>
>> I appreciate it if you can share your working sample config file with me.
>>
>> Thanks
>>
>> --
>>
>> _
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>>
>
>
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[asterisk-users] Anyone can share their config file for Cisco phone please?

[bruce bruce]

I have an *ipphone 7965G* which has to be connected to Asterisk. It has been
flashed with SIP firmware but the config file doesn't seem to work maybe I
am missing something in it.

I appreciate it if you can share your working sample config file with me.

Thanks
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[asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?

[bruce bruce]

Hi Everyone,

I am accustomed to PUTTY and it's very nice as in it allows many many SSH
profiles to be saved and allows tunneling etcbut it's not very good when
it comes to scrolling up and down, colors, text size, and specially it
doesn't give a title to the opened instance. Maybe giving the IP address as
the title of the window would help a lot if you have many different servers
opened at the same time.

Can you please weigh in and tell me what your favorite terminal software is
and why?

Thanks,
Bruce
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[asterisk-users] What are the guts of AsteriskNOW and how it compares to other popular flavors available?

[bruce bruce]

Hi Everyone,

I want to know a bit about the guts of the current AsterisNOW system. I know
that FreePBX is embraced as the main GUI but is just an install of CentOS
5.4 + (Asterisk/FreePBX from Yum repos)?

- Or is there anymore to this? Maybe some security tools?
- Or is Asterisk built from the source?
- Is there some other program installed to facilitate a better PBX system
than those of the competitor flavours (e.g. Trixbox, piaf, Elastix)? or does
it generally fall behind those?
- I would be very happy if you can point me to section of the image file
which is responsible for post install of CentOS as I think that is the
approach to get an AsteriskNOW system running with Asterisk and etc...

Thanks,
Bruce
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[asterisk-users] A lot of : doing dnsmgr_lookup for - Asterisk installed from YUM

[bruce bruce]

Hi Guys,

Asterisk 1.6.2.7 install from Yum Repository shows a lot of :> doing
dnsmgr_lookup for sip.provider.com

Google searches show it was fixed in some version.

Is this to be ignored?

Thanks
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Re: [asterisk-users] I look ARI (Asterisk Recording Interface)

[bruce bruce]

It's one of the bad modules that goes with FreePBX anyhow. The moment you go
over 3000 recordings you are already in trouble. It's about time someone
come up with a better moduel.

On Wed, Jun 23, 2010 at 11:05 AM, Mickael Monsieur <
mickael.monsi...@gmail.com> wrote:

> Hello,
> I look ARI (Asterisk Recording Interface)
> the publisher site is closed...
>
> http://www.littlejohnconsulting.com/ari
>
> Thank you,
> Mickael
>
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Re: [asterisk-users] Xorocom Missing files...where to get it? astribank_upgrade

[bruce bruce]

What about after step 3? That is where the messy instructions begin. I am
not trying to bash but I just had to resort to google to find  this which is
not included in the Trixbox 2.8 installation instructions:

wget http://svn.digium.com/svn/dahdi/tools/trunk/xpp/astribank_upgrade
chmod +x astribank_upgrade

There is a disconnect in steps and there is flow. That is what bugs me.
Anyone can find the support link on that page. I was talking about the
general status of installation instructions. I don't know why it is so hard
to do a 1,2,3 install and DONE, specially when they can separate Trixbox
from Elastix and version to version.


On Tue, Jun 22, 2010 at 4:53 PM, Steve Edwards wrote:

> On Tue, 22 Jun 2010, bruce bruce wrote:
>
> > I was on Xorocom site but there is no clear and consice place to
> > download drivers and firmware. I am reading their instructions to
> > install Astribank 8 channel FXO on Trixbox 2.8 and I seem to be missing
> > files at this step:
>
> [snip]
>
> > Where the heck are these files on their site?
> >
> > It's really a bugger when a manufacturer can't organize a site
> > nicelysigh
>
> Based on the professionalism I've always seen from Tzafrir this struck me
> as odd so I thought I'd take a look...
>
> What's hard about:
>
> 1) Hover over "Support"
>
> 2) Select "Upgrades & Downloads"
>
> 3) Click on "Astribank Drivers"
>
> Seemed pretty obvious to me. Am I missing something?
>
> --
> 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] Xorocom Missing files...where to get it? astribank_upgrade

[bruce bruce]

Hi Everyone,

I was on Xorocom site but there is no clear and consice place to download
drivers and firmware. I am reading their instructions to install Astribank 8
channel FXO on Trixbox 2.8 and I seem to be missing files at this step:

[pbx.archology.com dahdi]# /usr/share/doc/astribank_upgrade
/usr/share/dahdi/
-bash: /usr/share/doc/astribank_upgrade: No such file or directory


Where the heck are these files on their site?

It's really a bugger when a manufacturer can't organize a site
nicelysigh

Thanks guys
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Re: [asterisk-users] Dialplan Gurus? Can Asterisk 1.4x CHANNEL function be used to retrieve info about OTHER channels?

[bruce bruce]

Thanks for the input. If this is doable via Asterisk AMI why not through
dial-plan? I mean it only makes sense to be possible through dial-plan where
all access is given as well just like the AMI. Am I wrong with this?

On Tue, Jun 22, 2010 at 4:01 PM, Elliot Otchet <
elliot.otc...@callingcircles.com> wrote:

>  Get it via the AMI.  If you’re already using PHPAGI, it is trivial to get
> this data.  You can even find an example of how to call “sip show peers” and
> output the resulting response.  You avoid using the (-rx) and you get the
> data you were looking for.
>
>
>
> http://phpagi.sourceforge.net/phpagi2/docs/phpAGI/AGI_AsteriskManager.html
>
>
>
> If you’re already using PHPAGI often on a busy system, you
> might want to get more ram, use fastagi to move the PHP load to another
> system, or take Steve Edward’s standard advice and rewrite it in C.   thoughts>
>
>
>
> -Elliot
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce
> *Sent:* Tuesday, June 22, 2010 1:32 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Dialplan Gurus? Can Asterisk 1.4x CHANNEL
> function be used to retrieve info about OTHER channels?
>
>
>
> Thanks Tiago and Tzafrir. I agree with the heavy load that Tzafrir
> mentioned. I already made a phpagi that does a system() for asterisk -rx and
> it's not very responsive at time.
>
>
>
> So what is the solution guys?
>
>
>
> You see, I only want to know if g729 is being used because I want to
> determine if a trunk is being used or not. Now, don't be hasty and suggest
> GROUP_COUNT to me as I can not use that because I can only see the calls by
> "sip show peers" or "core show channels" and "group show channels" doesn't
> show me any channels because I do not have control over the calls place as
> they are placed by A2Billing.
>
>
>
> Any more Gurus want to weigh in more?
>
>
>
> On Tue, Jun 22, 2010 at 6:42 AM, Tzafrir Cohen 
> wrote:
>
> On Tue, Jun 22, 2010 at 11:25:29AM +0100, Tiago Geada wrote:
> > Hi!
> >
> > If it was me, I would create a bash script calling asterisk -vrx "core
> show
> > commands"
> >
> > something like:
> >
> > for chan in $(asterisk -vrx "core show channels concise");
> > do
> > asterisk -vrx "core show channel $(echo $chan|cut -d \! -f1)"|grep -i
> > native;
> > done
>
> The overhead of each 'asterisk -rx' command is noticable. If you have 10
> calls or more, this can have an odd effect.
>
> Not to mention that the fact that it is so slow exposes its raciness[1].
>
>
> >
> > On 21 June 2010 16:08, bruce bruce  wrote:
> >
> > > Hi Everyone,
> > >
> > > I want to know if a specific codec type is used at least one. For
> example,
> > > I want to know if out of the 100 calls on the system if there is a 1
> channel
> > > that is running G.729 codec right now. If using dial-plan and I dial
> in, I
> > > can use this to obtain information about CURRENT channel. But it won't
> allow
> > > me to obtain information about OTHER channels and that is what I want
> to do.
> > > I want a search for all channels and an output spit out as g729 or TRUE
> or
> > > FALSE if there is a g729 channel.
> > >
> > > exten => s,1,Answer()
> > > exten => s,n,Set(foo=${CHANNEL(audioreadformat)})
> > > exten => s,n,NoOp(${foo})
> > >
> > > Above  NoOp spits out g729 if I call in with a g729 codec. But I
> want that to be about other channels and not the one I am calling into.
> > >
> > > Thanks,
> > >
> > > Bruce
>
> [1] Which should naturally be fixed using locks :-)
>
>
> --
>   Tzafrir Cohen
> icq#16849755  
> jabber:tzafrir.co...@xorcom.com
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
>
> --
> _
>
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
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> This message is intended only for the use of the individual (s) or entity
>

Re: [asterisk-users] Dialplan Gurus? Can Asterisk 1.4x CHANNEL function be used to retrieve info about OTHER channels?

[bruce bruce]

Thanks Tiago and Tzafrir. I agree with the heavy load that Tzafrir
mentioned. I already made a phpagi that does a system() for asterisk -rx and
it's not very responsive at time.

So what is the solution guys?

You see, I only want to know if g729 is being used because I want to
determine if a trunk is being used or not. Now, don't be hasty and suggest
GROUP_COUNT to me as I can not use that because I can only see the calls by
"sip show peers" or "core show channels" and "group show channels" doesn't
show me any channels because I do not have control over the calls place as
they are placed by A2Billing.

Any more Gurus want to weigh in more?


On Tue, Jun 22, 2010 at 6:42 AM, Tzafrir Cohen wrote:

> On Tue, Jun 22, 2010 at 11:25:29AM +0100, Tiago Geada wrote:
> > Hi!
> >
> > If it was me, I would create a bash script calling asterisk -vrx "core
> show
> > commands"
> >
> > something like:
> >
> > for chan in $(asterisk -vrx "core show channels concise");
> > do
> > asterisk -vrx "core show channel $(echo $chan|cut -d \! -f1)"|grep -i
> > native;
> > done
>
> The overhead of each 'asterisk -rx' command is noticable. If you have 10
> calls or more, this can have an odd effect.
>
> Not to mention that the fact that it is so slow exposes its raciness[1].
>
> >
> > On 21 June 2010 16:08, bruce bruce  wrote:
> >
> > > Hi Everyone,
> > >
> > > I want to know if a specific codec type is used at least one. For
> example,
> > > I want to know if out of the 100 calls on the system if there is a 1
> channel
> > > that is running G.729 codec right now. If using dial-plan and I dial
> in, I
> > > can use this to obtain information about CURRENT channel. But it won't
> allow
> > > me to obtain information about OTHER channels and that is what I want
> to do.
> > > I want a search for all channels and an output spit out as g729 or TRUE
> or
> > > FALSE if there is a g729 channel.
> > >
> > > exten => s,1,Answer()
> > > exten => s,n,Set(foo=${CHANNEL(audioreadformat)})
> > > exten => s,n,NoOp(${foo})
> > >
> > > Above  NoOp spits out g729 if I call in with a g729 codec. But I
> want that to be about other channels and not the one I am calling into.
> > >
> > > Thanks,
> > >
> > > Bruce
>
> [1] Which should naturally be fixed using locks :-)
>
> --
>   Tzafrir Cohen
> icq#16849755  
> jabber:tzafrir.co...@xorcom.com
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
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Re: [asterisk-users] Can't make calls out of Astribank - show regdump doesn't show voltage - Getting warning due to old driver

[bruce bruce]

Thanks for the tips.

This is an 8 FXO channel Astribank. My understanding is that Trixbox 2.8
already had everything dahdi related installed so there is no driver from
Astribank. I followed this page and did rpm -Uvh for freepbx-module-zapauto

http://www.xorcom.com/downloads/astribank2-dahdi.html

What other steps do I have to take to complete the installation if you think
I have not finished. Or where can I look to find the problem?

As you could see from my last e-mail, everything seemed fine except for the
voltage part. I am SURE that PSTN lines are connected to the box.

Thanks

<http://www.xorcom.com/downloads/astribank2-trixbox-ce-drivers.html#trixboxce2.8>

On Tue, Jun 22, 2010 at 5:25 AM, Tzafrir Cohen wrote:

> On Tue, Jun 22, 2010 at 12:58:00AM -0400, bruce bruce wrote:
> > Hi Guys,
> >
> > An 8 channel
>
> FXO?
>
> > Astribank is connected to Trixbox 2.8 and I ran
> > freepbx-module-zapauto but I get the following when running these
> > commands and can't make calls out:
> >
> > [Trixbox]# dahdi_genconf xpporder
> > /usr/sbin/dahdi_genconf: warning - OLD DRIVER: missing
> > '/sys/bus/xpds/devices/00:0:0/timing_priority'. Fall back to /proc
>
> Which one, exactly? Trixbox originally had rather dates DAHDI drivers. I
> believe you should now be able to find much newer ones. At least in
> their repos.
>
> >
> > pbx*CLI> dahdi show channels
> > Chan Extension Context Language MOH Interpret Blocked State
> >
> > pseudo default default In Service
> > 1 from-pstn default In Service
> > 2 from-pstn default In Service
> > 3 from-pstn default In Service
> > 4 from-pstn default In Service
> > 5 from-pstn default In Service
> > 6 from-pstn default In Service
> >
> > 7 from-pstn default In Service
> > 8 from-pstn default In Service
> >
> > pbx*CLI> dahdi show status
> > Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO
> > Xorcom XPD #00/00: FXO OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1)
> >
> >
> > pbx*CLI> dahdi show regdump 1
> > Unable to get registers on channel 1
> > Unable to get stats on channel 1
>
> I believe that regdump uses some specific interface to the Digium card.
> If you want a bunch of technical information you don't really
> understand, look under /proc/xpp/ . I'm not going to start explaining it
> beyond
>
> http://docs.tzafrir.org.il/dahdi-linux/README.Astribank.html#_proc_interface
>
> >
> > [Trixbox]# dahdi_hardware -v
> > /usr/sbin/dahdi_hardware: warning - OLD DRIVER: missing
> > '/sys/bus/xpds/devices/00:0:0/timing_priority'. Fall back to /proc
> >
> > usb:002/002 xpp_usb+ e4e4:1162 Astribank-modular FPGA-firmware
> > LABEL=[usb:X1044207] connect...@usb-:00:1d.7-4
> > XBUS-00/XPD-00: FXO (8) Span 1 DAHDI-SYNC
> >
> >
> > Note: This Astribank is deployed in United Arab Emirates and I am not
> > sure what the line type is in terms of Ground or Loop start and
> > wondering if that makes a difference with the Astribank and the fact
> > that it can't how the voltage using "show regdump"
>
> IIRC they use LS (That is: no power denial is used at the end of a
> call).
>
> >
> > And I am definitly not sure what that warning of OLD DRIVER is about.
> > Any help is appreciated.
>
> At the time we wrote it, we relied heavily on procfs. However procfs is
> not something that can be safely used when the module is loaded or
> removed. This is a fine way to get panics. Thus we gradually moved many
> things from /proc to /sys .
>
> Typically such a message would mean a combination of older dahdi-linux
> (loaded) and newer dahdi-tools. Though it's a warning, and not an error.
>
> --
>   Tzafrir Cohen
> icq#16849755  
> jabber:tzafrir.co...@xorcom.com
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
>
> --
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[asterisk-users] Can't make calls out of Astribank - show regdump doesn't show voltage - Getting warning due to old driver

[bruce bruce]

Hi Guys,

An 8 channel Astribank is connected to Trixbox 2.8 and I ran
freepbx-module-zapauto but I get the following when running these
commands and can't make calls out:

[Trixbox]# dahdi_genconf xpporder
/usr/sbin/dahdi_genconf: warning - OLD DRIVER: missing
'/sys/bus/xpds/devices/00:0:0/timing_priority'. Fall back to /proc

pbx*CLI> dahdi show channels
Chan Extension Context Language MOH Interpret Blocked State

pseudo default default In Service
1 from-pstn default In Service
2 from-pstn default In Service
3 from-pstn default In Service
4 from-pstn default In Service
5 from-pstn default In Service
6 from-pstn default In Service

7 from-pstn default In Service
8 from-pstn default In Service

pbx*CLI> dahdi show status
Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO
Xorcom XPD #00/00: FXO OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1)


pbx*CLI> dahdi show regdump 1
Unable to get registers on channel 1
Unable to get stats on channel 1

[Trixbox]# dahdi_hardware -v
/usr/sbin/dahdi_hardware: warning - OLD DRIVER: missing
'/sys/bus/xpds/devices/00:0:0/timing_priority'. Fall back to /proc

usb:002/002 xpp_usb+ e4e4:1162 Astribank-modular FPGA-firmware
LABEL=[usb:X1044207] connect...@usb-:00:1d.7-4
XBUS-00/XPD-00: FXO (8) Span 1 DAHDI-SYNC


Note: This Astribank is deployed in United Arab Emirates and I am not
sure what the line type is in terms of Ground or Loop start and
wondering if that makes a difference with the Astribank and the fact
that it can't how the voltage using "show regdump"

And I am definitly not sure what that warning of OLD DRIVER is about.
Any help is appreciated.


Thanks,
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[asterisk-users] Dialplan Gurus? Can Asterisk 1.4x CHANNEL function be used to retrieve info about OTHER channels?

[bruce bruce]

Hi Everyone,

I want to know if a specific codec type is used at least one. For example, I
want to know if out of the 100 calls on the system if there is a 1 channel
that is running G.729 codec right now. If using dial-plan and I dial in, I
can use this to obtain information about CURRENT channel. But it won't allow
me to obtain information about OTHER channels and that is what I want to do.
I want a search for all channels and an output spit out as g729 or TRUE or
FALSE if there is a g729 channel.

exten => s,1,Answer()
exten => s,n,Set(foo=${CHANNEL(audioreadformat)})
exten => s,n,NoOp(${foo})

Above  NoOp spits out g729 if I call in with a g729 codec. But I
want that to be about other channels and not the one I am calling
into.

Thanks,

Bruce
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[asterisk-users] Deleting some of the CDR data - How to do it safely?

[bruce bruce]

Hi Guys,

I am looking to delete some of the CDR logged by Asterisk in asteriskcdrdb
in a PbxinaFlash system running Asterisk 1.4.x

The CDR records to deleted are probably a big chunk and spread out all
through the database but I basically want to delete all calls that came in
through a specific DID. I think they all show as SIP/did_number

I don't want to break the system or break the reporting tool of FreePBX so
some advice would be appreciated.

Also, any MySQL commands related to deleting such records with conditions
above mentioned would be helpful.

Thanks
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Re: [asterisk-users] Using SetVar with System() is it possible?

[bruce bruce]

I am using Asterisk v1.4.x

I have got php along with phpagi to report back using a system call within
php to asterisk -rx but it's buggy. For one thing php is hanging sometimes
and not returning anything.

Wondering if I can run search to see if a g.729 channel is up or not using
only the dial-plan. I want to get a TRUE or FALSE in a var for it.

I was pointed to GROUP_COUNT but that doesn't cut it for me in this case.

There is gotta be some simple dial-plan to find out if a trunk is being used
or if a g.729 call is going through or if a specific destination is being
dialed with first three digits being 789 (for example) as all of those
variables can help me move to next step which is to decide to place a second
call through the same trunk or not.

Any inputs?

Thanks a lot



On Sat, Jun 19, 2010 at 1:56 PM, Tzafrir Cohen wrote:

> On Sat, Jun 19, 2010 at 10:58:17AM -0400, bruce bruce wrote:
> > Hi Guys,
> >
> > Is it possible to harvest the output of system into a SetVar(variable)?
> >
> > exten => s,n,SetVar(var=system(*asterisk -rx "sip show channels" | grep
> -c
> > "(ulaw)")*
>
> There's the function SHELL. Though I suspect you use 1.2 and I'm not
> sure if it was there already.
>
> Anyway, I'm the output of that command is so simple to parse.
>
> --
>   Tzafrir Cohen
> icq#16849755  
> jabber:tzafrir.co...@xorcom.com
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
>
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[asterisk-users] Using SetVar with System() is it possible?

[bruce bruce]

Hi Guys,

Is it possible to harvest the output of system into a SetVar(variable)?

exten => s,n,SetVar(var=system(*asterisk -rx "sip show channels" | grep -c
"(ulaw)")*
*
*
*??? any problem with the syntax? *
*
*
*
*
*Thanks,*
*
*
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Re: [asterisk-users] asterisk issue

[bruce bruce]

Nice and colorful tutorial for cronjobs.
http://www.linuxconfig.org/Linux_Cron_Guide

-Bruce

On Fri, Jun 18, 2010 at 1:55 PM, salaheddine elharit <
salah.elharit...@gmail.com> wrote:

> thanks for your response
>
> how can i create and execute this cron
>
> 2010/6/18 Danny Nicholas 
>
>>  I do a cron to execute “/usr/sbin/asterisk –rx “restart when convenient”
>> “ each day at 4:45 AM.  This doesn’t really “solve” any problems, just does
>> “housekeeping” to keep a clean environment, since some installs/os’es lend
>> themselves to memory leaks.
>>
>>
>>  --
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
>> elharit
>> *Sent:* Friday, June 18, 2010 5:13 AM
>> *To:* asterisk-users@lists.digium.com
>> *Subject:* [asterisk-users] asterisk issue
>>
>>
>>
>> Hello,
>>
>>
>>
>> I have a problem in Asterisk 1.4 each day I need to restart *asterisk
>> service asterisk* restart in order to unblock the calls
>>
>> My question how can I do in order to check the issue, and if there is any
>> tool or log?
>>
>>
>>
>> Thanks and regards.
>>
>> --
>>
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>   http://www.asterisk.org/hello
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Re: [asterisk-users] How to pass variable back and forth from dialplan to php file?

[bruce bruce]

I do have a rather bigger project coming my way and I would really like to
know how to do the *"feeding a text file into the STDIN of your AGI so you
can debug completely external to Asterisk"*
*
*
*However, for this project, it seems that I can use php system() along with
grep to see the status of a peer with one line of code:*
*
*
*
 asterisk -rx "sip show peer $sip_peer" | grep -c "X-Lite"'
Above ^^^ in Linux prompt returns 1 if $sip_peer is registered with X-Lite
else it returns 0.

But using system() I think the "" confuse php and value is always returned
as 0.
$peer_count = system('asterisk -rx "sip show peer $sip_peer" | grep -c
"X-Lite"', $retval);

Should $sip_peer be inside another set of parenthesis?

Thanks,
Bruce

*
On Mon, Jun 14, 2010 at 6:44 PM, Steve Edwards wrote:

> On Mon, 14 Jun 2010, bruce bruce wrote:
>
>  Thanks for the input. I actually did use verbose() and that is when I
>> noticed that my path to phpagi was not right since nothing was coming
>> through. For return value prior to fixing phpagi path, I was getting:
>>
>> NoOp("SIP/64.111.222.111-0ca7", "")
>>
>> which actually is just right because if you notice the last three
>> charecters of that line is ""). So, when the phpagi path is correct, it
>> looks like:  "415444555").
>>
>
> Which is why enabling AGI debugging and posting actual console code can
> speed problem resolution. Sometimes it's just another set of eyes on the
> same output :)
>
> If you're going to be developing a significant bit of AGI code, there is a
> "not immediately obvious" technique of feeding a text file into the STDIN of
> your AGI so you can debug completely external to Asterisk -- even within an
> IDE like emacs+gdb. (Or whatever the PHP equivalence is.)
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
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Re: [asterisk-users] How to pass variable back and forth from dialplan to php file?

[bruce bruce]

Thanks for the input. I actually did use verbose() and that is when I
noticed that my path to phpagi was not right since nothing was coming
through.

For return value prior to fixing phpagi path, I was getting:

*NoOp("SIP/64.111.222.111-0ca7", "")*

which actually is just right because if you notice the last three charecters
of that line is* "")*. So, when the phpagi path is correct, it looks like:
*"415444555")*.

-Bruce

On Mon, Jun 14, 2010 at 6:09 PM, Steve Edwards wrote:

> > On Mon, 2010-06-14 at 14:57 -0400, bruce bruce wrote:
>
> >> Carlos, Thanks a lot for getting me started. That helps a great deal.
>
> >> exten => _x.,1,NoOp(${EXTEN})
>
> Since you're just getting started, there is an application specifically
> written to send output to the CLI -- verbose(). It's more "obvious" and it
> has additional functionality.
>
> On Mon, 14 Jun 2010, Carlos Chavez wrote:
>
> > As you can see agi_extension and agi_dnid should contain the number that
> > the user dialed.  Those are all the variables that are automatically
> > sent to the AGI script from Asterisk.
>
> agi_extension contains the extension (the bit after the equals sign) of
> the statement that executes the AGI application. For example,
>
>exten = s,n,agi(foo)
>
> When "foo" examines agi_extension, it will contain "s"
>
> > I do not know why you are getting the channel instead of the extension,
> > you could try giving the extension as a parameter to the AGI script if
> > you cannot get that from the included request variable.
>
> Better to fix the really simple stuff before he gets to the complex stuff.
>
> --
> 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] Small PC for Asterisk appliance to support 2 x Sangoma A200 (2 x PCIe standard cards)

[bruce bruce]

Hi Guys,

Looking for a powerful box that is compact, can take two hard drives for
Raid-1 (no SSD, too expensive), have at least two Gig ports or two
10/100mbps ports. Fit two PCIe or one PCIe card plus it's daughter card
which needs as much room as a PCIe and doesn't need the actual slot. That is
for Sangoma A200 + Daughter card.

I would be really glad if I can get some sort of a KVM over IP for this box
built in as this is going to be deployed overseas.

One of these servers is to support 30 extensions with transacoding g.729 and
the other is to support 10 extensions with transcoding. I don't think there
is anything in between in terms of hardware but any and all suggestions are
welcome.

Remember: Compact, quite, good airflow, powerful, 2 x empty slots one of
which is a PCIe and 2xNIC ports.

Thanks
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Re: [asterisk-users] How to pass variable back and forth from dialplan to php file?

[bruce bruce]

Thanks for pointing to the debug. I found that the path to phpagi was not
right and since I fixed that everything seems to work fine.

Now, I want to know if I can use a phpagi command to check the status of SIP
Peer if it is online and registered or not. I know I can use *grep with
asterisk rx "sip show peers" *and use that as a shell script but I think
there are better methods in Asterisk dialplan or in phpagi that it can be
check.

Thanks,
Bruce

On Mon, Jun 14, 2010 at 5:27 PM, Carlos Chavez wrote:

>If you do an "agi set debug on" from the CLI and run your AGI you
> should see something like this:
>
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_request: auth.agi
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_channel: SIP/206-0d9b
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_language: es
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_type: SIP
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_uniqueid: 1276192380.5115
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_version: 1.6.2.8
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_callerid: 206
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_calleridname: Carmen Saavedra
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_callingpres: 0
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_callingani2: 0
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_callington: 0
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_callingtns: 0
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_dnid: *2208101282711
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_rdnis: unknown
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_context: oficina
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_extension: *2208101282711
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_priority: 2
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_enhanced: 0.0
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_accountcode:
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_threadid: -1234945136
> [Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
> agi_arg_1: ldnac
>
>As you can see agi_extension and agi_dnid should contain the number
> that the user dialed.  Those are all the variables that are
> automatically sent to the AGI script from Asterisk.  I do not know why
> you are getting the channel instead of the extension, you could try
> giving the extension as a parameter to the AGI script if you cannot get
> that from the included request variable.
>
> On Mon, 2010-06-14 at 14:57 -0400, bruce bruce wrote:
> > Carlos, Thanks a lot for getting me started. That helps a great deal.
> >
> >
> > Currently, the $agi->request['agi_extension'];  returns the SIP
> > channel info with IP and I want that to be the incoming DID number.
> >
> >
> > My dialplan output is this for line one:
> >
> >
> > exten => _x.,1,NoOp(${EXTEN})
> >
> >
> > 415444555
> >
> >
> > But with the agi_extension it comes back as:
> > NoOp("SIP/64.111.222.111-0ca7", "")
> >
> >
> > Where can I find the list of command requests that can be sent to
> > Asterisk? Specially that for DID.
> >
> >
> > Thanks
> >
> >
> > On Mon, Jun 14, 2010 at 2:15 PM, Carlos Chavez
> >  wrote:
> > On Mon, 2010-06-14 at 13:41 -0400, bruce bruce wrote:
> > > Hi Carlso,
> > >
> > >
> > > Thanks for the input. I have done this in php and am not
> > familiar with
> > > phpagi.
> > > So, there is absolutely no way to temporarily solve this
> > problem by
> > > getting the value back from php file?
> > >
> > >
> > > Wondering if it would require a lot of work to change the
> > php file to
> > > phpagi?
> > > Thanks,
> > > Bruce
> > >
> >
> >Here is an example:
> >
> > exten => _x.,1,AGI(sample.agi)
> > exten => _x.,n,NoOp(${var})
> >
> >
> > sample.agi:
> >
> > #!/usr/bin/php -q
> >  > set_time_limit(30);
&

Re: [asterisk-users] How to pass variable back and forth from dialplan to php file?

[bruce bruce]

Hi again,

So, I have this but NoOp shows a random SIP info value rather then the one
passed to it. Just to test, I am sending $didin as argument to test.php and
then expect it back as $didgot back into dialplan. But it seems that either
send or receive has problem because no matter what I put as the NoOp in _x,4
it all comes back the same as SIP channel info.

*[custom-phpagi]*
*exten => _x.,1,Set(didin=${EXTEN})*
*exten => _x.,2,NoOp(${didin}*
*exten => _x.,3,deadAGI(test.agi,${didin})*
*exten => _x.,4,NoOp(${didgot})*
*exten => _x.,5,Hangup()*

*/var/lib/asterisk/agi-lib/test.agi:*

#!/usr/bin/php -q
get_variable("didgot", $didgot);  /// both of these don't seem to
send didgot back to dialplan
$agi->set_variable("didgot", $didgot);  /// both of these don't seem to
send didgot back to dialplan


exit(0);
?>

Thanks,
Bruce


On Mon, Jun 14, 2010 at 2:15 PM, Carlos Chavez wrote:

> On Mon, 2010-06-14 at 13:41 -0400, bruce bruce wrote:
> > Hi Carlso,
> >
> >
> > Thanks for the input. I have done this in php and am not familiar with
> > phpagi.
> > So, there is absolutely no way to temporarily solve this problem by
> > getting the value back from php file?
> >
> >
> > Wondering if it would require a lot of work to change the php file to
> > phpagi?
> > Thanks,
> > Bruce
> >
> Here is an example:
>
> exten => _x.,1,AGI(sample.agi)
> exten => _x.,n,NoOp(${var})
>
> sample.agi:
>
> #!/usr/bin/php -q
>  set_time_limit(30);
> require('phpagi/phpagi.php');
> $agi = new AGI();
> $exten = $agi->request['agi_extension']; //Dialed extension
> // the result is stored in $exten
> // do something with your data
> $agi->set_variable("var", $result);
> $agi->verbose("The result was: $result", 3);
> ?>
>
>You can even send parameters to the AGI via the command line like:
>
> exten => _x.,1,AGI(sample.agi,param1,param2) //Use comma for 1.6 or |
> for 1.4 or below
>
>And access them via $argv[1], $argv[2] is there is some extra
> information that you cannot get via the AGI variables.
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez Prats
> Director de Tecnología
> +52-55-91169161 ext 2001
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] How to pass variable back and forth from dialplan to php file?

[bruce bruce]

Carlos, Thanks a lot for getting me started. That helps a great deal.

Currently, the *$agi->request['agi_extension'];  returns the SIP channel
info with IP and I want that to be the incoming DID number. *
*
*
*My dialplan output is this for line one:*

**exten => _x.,1,NoOp(${EXTEN})

*415444555*

But with the agi_extension it comes back as:
NoOp("SIP/64.111.222.111-0ca7", "")

Where can I find the list of command requests that can be sent to Asterisk?
Specially that for DID.

Thanks


On Mon, Jun 14, 2010 at 2:15 PM, Carlos Chavez wrote:

> On Mon, 2010-06-14 at 13:41 -0400, bruce bruce wrote:
> > Hi Carlso,
> >
> >
> > Thanks for the input. I have done this in php and am not familiar with
> > phpagi.
> > So, there is absolutely no way to temporarily solve this problem by
> > getting the value back from php file?
> >
> >
> > Wondering if it would require a lot of work to change the php file to
> > phpagi?
> > Thanks,
> > Bruce
> >
> Here is an example:
>
> exten => _x.,1,AGI(sample.agi)
> exten => _x.,n,NoOp(${var})
>
> sample.agi:
>
> #!/usr/bin/php -q
>  set_time_limit(30);
> require('phpagi/phpagi.php');
> $agi = new AGI();
> $exten = $agi->request['agi_extension']; //Dialed extension
> // the result is stored in $exten
> // do something with your data
> $agi->set_variable("var", $result);
> $agi->verbose("The result was: $result", 3);
> ?>
>
>You can even send parameters to the AGI via the command line like:
>
> exten => _x.,1,AGI(sample.agi,param1,param2) //Use comma for 1.6 or |
> for 1.4 or below
>
>And access them via $argv[1], $argv[2] is there is some extra
> information that you cannot get via the AGI variables.
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez Prats
> Director de Tecnología
> +52-55-91169161 ext 2001
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] How to pass variable back and forth from dialplan to php file?

[bruce bruce]

Hi Carlso,

Thanks for the input. I have done this in php and am not familiar with
phpagi.
So, there is absolutely no way to temporarily solve this problem by getting
the value back from php file?

Wondering if it would require a lot of work to change the php file to
phpagi?
Thanks,
Bruce

On Mon, Jun 14, 2010 at 12:12 PM, Carlos Chavez wrote:

> On Mon, 2010-06-14 at 12:00 -0400, bruce bruce wrote:
> > Hi Everyone,
> >
> >
> > I have a php file that if an argument is passed to it, it will echo a
> > number back. I am looking to use system() in dial-plan to send
> > ${EXTEN} to it and then to get that processed value back from the php
> > file and put it in $var back into asterisk dial-plan. While trying
> > this method doesn't work:
> >
> >
> > exten => _x.,1,SetVar(var = system(php /file.php ${EXTEN})
> > exten => _x.,n,NoOp(${var})
> >
> >
>
> This is exactly what AGI was made for.  Look into the PHPAGI
> library.
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez Prats
> Director de Tecnología
> +52-55-91169161 ext 2001
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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[asterisk-users] How to pass variable back and forth from dialplan to php file?

[bruce bruce]

Hi Everyone,

I have a php file that if an argument is passed to it, it will echo a number
back. I am looking to use system() in dial-plan to send ${EXTEN} to it and
then to get that processed value back from the php file and put it in $var
back into asterisk dial-plan. While trying this method doesn't work:

exten => _x.,1,SetVar(var = system(php /file.php ${EXTEN})
exten => _x.,n,NoOp(${var})

What is right syntax for line 1?

Thanks,
Bruce
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[asterisk-users] Am I having problems with codecs? or am I not receiving an invite at all from my DID provider?

[bruce bruce]

Hi Guys,

I have Spikko setup as provider of DID and outbound routes and I can make
calls out but no inbound calls via DID can be made. I did a sip debug which
is reported below. I never receive the call though, I have a catch all in my
inbound routes and it doesn't hit my context at all or not sip invite comes
in:


FreePBX:

Trunk Name:
*Spikko*

Peer Detail
*username=MyUsername*
*type=friend*
*secret=MyPassword*
*host=sip.spikko.com*
*nat=no*
*port=5090*
*fromuser=MyUsername*
*disallow=all*
*allow=g729&gsm&ulaw&alaw*

Register String:
*MyUsername:mypassw...@sip.spikko.com:5090/MyUsername*


Inbound Router:
*Send Any DID and ANY CID to Music on Hold*


Sip debug:

*Really destroying SIP dialog '
417b3c8f3a97a82d4629343a53b2f...@177.177.177.177' Method: REGISTER*
*tel*CLI>*
*<--- SIP read from UDP:82.80.252.29:5090 --->*
*INVITE sip:myusern...@177.177.177.177 SIP/2.0
*
*Via: SIP/2.0/UDP 82.80.252.234:5090;branch=z9hG4bK07b38a0c;rport*
*From: "Unknown" ;tag=as24089849*
*To: >*
*Contact: *
*Call-ID: 55a4cf1f4e5575e97f8b3b23495f0...@82.80.252.234*
*CSeq: 102 INVITE*
*User-Agent: AG1*
*Max-Forwards: 70*
*Date: Thu, 10 Jun 2010 14:58:09 GMT*
*Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY*
*Supported: replaces*
*Content-Type: application/sdp*
*Content-Length: 331*
*
*
*v=0*
*o=root 6129 6129 IN IP4 82.80.252.234*
*s=session*
*c=IN IP4 82.80.252.234*
*t=0 0*
*m=audio 10172 RTP/AVP 18 3 97 101*
*a=rtpmap:18 G729/8000*
*a=fmtp:18 annexb=no*
*a=rtpmap:3 GSM/8000*
*a=rtpmap:97 iLBC/8000*
*a=fmtp:97 mode=30*
*a=rtpmap:101 telephone-event/8000*
*a=fmtp:101 0-16*
*a=silenceSupp:off - - - -*
*a=ptime:20*
*a=sendrecv*
*
*
*<->*
*--- (14 headers 16 lines) ---*
*Using INVITE request as basis request -
55a4cf1f4e5575e97f8b3b23495f0...@82.80.252.234*
*Found peer 'Spikko' for 'Unknown' from 82.80.252.29:5090*


I also sometimes get this even though trunk shows registered and can make
calls out:
*<--- Transmitting (no NAT) to 82.80.252.29:5090 --->*
*SIP/2.0 489 Bad event*
*Via: SIP/2.0/UDP 82.80.252.234:5090
;branch=z9hG4bK463b703d;received=82.80.252.29;rport=5090*
*From: "asterisk" ;tag=as4af8cf81*
*To: 
>;tag=as64c0ba34*
*Call-ID: 497197a679122f5d448d324f571f3...@82.80.252.234*
*CSeq: 102 NOTIFY*
*Server: Asterisk PBX 1.6.2.7*
*Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO*
*Supported: replaces, timer*
*Content-Length: 0*

Thanks,
Bruce
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Re: [asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk

[bruce bruce]

I actually commented all this in safe_asterisk and now asterisk loads all
fine at the beginning. Is this okay to do? Also at the beginning of the file
I commented #TTY=9 as well. Can someone shade some light as to what TTY is
and how it can have an adverse effect if it's not available?

#if test "x$TTY" != "x" ; then
#   if test -c /dev/tty${TTY} ; then
#   TTY=tty${TTY}
#   elif test -c /dev/vc/${TTY} ; then
#   TTY=vc/${TTY}
#   else
#   message "Cannot find specified TTY (${TTY})"
#   exit 1
#   fi
#   ASTARGS="${ASTARGS} -vvvg"
#   if test "x$CONSOLE" != "xno" ; then
#   ASTARGS="${ASTARGS} -c"
#   fi
#fi


On Mon, Jun 7, 2010 at 8:12 PM, bruce bruce  wrote:

> I did see the TTY=9 on the third or fourth line but commenting that doesn't
> help much. I would really appreciate it if you can send the changes you
> made.
>
> Indeed it is a VPS.
>
> Thanks,
> Bruce
>
> On Mon, Jun 7, 2010 at 7:49 PM, Warren Selby wrote:
>
>>
>> *chown: cannot access `/dev/tty9': No such file or directory*
>>
>>
>> I had this error on a VPS (virtual server) that did not have access to
>> tty's. You can take the TTY statement out of safe_asterisk script and then
>> try it again.  I don't have the exact code right now because I'm on my
>> phone, but you should be able to find it if you read through that file.
>>
>> Thanks,
>> --Warren Selby
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>   http://www.asterisk.org/hello
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Re: [asterisk-users] Deleting extension makes it usable?

[bruce bruce]

Since you mentioned FreePBX, unfortunately, it's not only the GUI that
drives the system and it can be that at some point someone planted
the extension in one of your .conf or other file if they had access to SSH
or some other way.

Going back to occurrence in sip.conf as mentioned, of course
FreePBX regenerates sip.conf every time and you can't tamper with it  but
sip_custom.conf or any other file can be called to just create
an extension in the non-GUI section and it will still work and not show up
in FreePBX GUI.

Recreating the extension probably over-wrote that or maybe supersedes that
and hence the failed authentication attempts.

If you can live with no SIP from outside, temporarily block any incoming on
5060 and 1-2.

To find the extension occurances in the .conf files, try this:

=> Delete the extension from FreePBX first and then:
cd /etc/asterisk
grep -o "3799" *.*

However, I think it's also possible to have the 3799 created in a totally
different directory in your server as long as it has the right
asterisk.asterisk permissions and it can be called by an #include from
sip_custom.conf. So, check that file out extensively.

-Bruce



On Tue, Jun 8, 2010 at 10:56 AM, Steve Murphy  wrote:

> I hope I'm correct, I don't have time to verify every bit of this,
> but
>
> The message
>
> [Jun  7 17:04:16] NOTICE[7422] chan_sip.c: Failed to authenticate user
> "asterisk" 
> >;tag=as23bacb61
>
> indicates the user "asterisk". Do you have a sip account for "asterisk"?
>
> Why it would take 14 seconds and an ANSWERED dial for an unathenticated
> use is something to investigate!
>
> As to the more general question of how 3799 could be unexpectedly matched
> in the dialplan, I would respond that there are several possibilities...
>
> One is, Is the account with the weak
> pword removed from sip.conf? The 3799 account? Because, it looks like
> SIP/206.20... (you abbreviated here in the CDR you listed) is where
> the call is originating.
>
> b. Did you *really* get rid of all 3799 occurrences in the dialplan? What
> patterns
> do you have in the dialplan that might match 3799, after the explicit 3799
> is removed?
> Any _ type patterns included or in the context in question?
>
> c. I uncovered a pattern matching bug, and reported it in bug
> https://issues.asterisk.org/view.php?id=17366
> where unexpected patterns are matched. Sorry, I haven't had time to correct
> it myself, it's probably
> a simple 1-line fix, but oh, what it might take to figure out what the line
> should say, and where it is!
>
> d. "s" is the "start" extension, and an incoming call will tend to get
> routed into an "s" extension.
>
> You can quickly determine (b) or (c), by going to the CLI, and saying
> "dialplan show 3...@whatever-context and see what turns up.
>
> murf
>
>
>
>
>
> On Tue, Jun 8, 2010 at 7:50 AM, J  wrote:
>
>> I'm fairly new to FreePBX/Asterisk/Trixbox, but have Googled myself
>> into submission here, so any assistance is appreciated.
>>
>> We had a user with a weak SIP secret recently that allowed it to be
>> used by an outside user. The extension was 3799. I could see the
>> intruder's calls (including the destination phone numbers) in the
>> trixbox call report log. Because the extension was no longer used, I
>> went ahead and deleted it, thinking that would solve the problem. I
>> also discovered approximately the same time that the Asterisk Call
>> Manager port was open to the outside world, which has since been
>> closed. The web interface, ssh, etc. have never been exposed to the
>> outside world. Since taking these actions, I restarted the asterisk
>> server.
>>
>> Now, here's the issue. I don't think deleting the extension helped.
>> Now I see entries like this in the reports log:
>>
>> Calldate  Channel Source Clid Dst Disposition Duration
>> 1.  2010-06-07 16:47:38 SIP/206.20...   3799"asterisk"
>> <3799>   s   ANSWERED00:14
>>
>> The "Dst" field being "s", where it used to be the phone number being
>> dialed. How is this extension able to be used even after it has been
>> deleted?
>>
>> Strangely, what I've done to keep the user out in the mean time is
>> re-created the 3799 extension with a better secret. This results in
>> log entries like the following:
>>
>> [Jun  7 17:04:16] NOTICE[7422] chan_sip.c: Failed to authenticate user
>> "asterisk" 
>> >;tag=as23bacb61
>>
>> Why can sip:3799 connect and make calls when the extension doesn't
>> exist? Is this person somehow using a "user" account? I've checked
>> both /etc/asterisk and the MySQL tables and am not coming up with
>> much. What does it mean that their destination is "s", not a phone
>> number?
>>
>> Thanks for any assistance!
>> J
>>
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Re: [asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk

[bruce bruce]

I did see the TTY=9 on the third or fourth line but commenting that doesn't
help much. I would really appreciate it if you can send the changes you
made.

Indeed it is a VPS.

Thanks,
Bruce

On Mon, Jun 7, 2010 at 7:49 PM, Warren Selby  wrote:

>
> *chown: cannot access `/dev/tty9': No such file or directory*
>
>
> I had this error on a VPS (virtual server) that did not have access to
> tty's. You can take the TTY statement out of safe_asterisk script and then
> try it again.  I don't have the exact code right now because I'm on my
> phone, but you should be able to find it if you read through that file.
>
> Thanks,
> --Warren Selby
>
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Re: [asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk

[bruce bruce]

Thanks for the input Seann and Steve. That is insightful. I did run
chkconfig --list asterisk and following is the output:
*[r...@tel ~]# chkconfig --list asterisk*
*asterisk0:off   1:off   2:on3:on4:on5:on6:off*

In file /usr/sbin/safe_asterisk I have priority for asterisk set at 9.
*# run asterisk with this priority*
*PRIORITY=9*

/var/log/messages doesn't show anything important or related to why asterisk
not starting at startup. I think asterisk should start first and then
amportal will start as well is asterisk does start.

Here is what happens if I do amportal restart:

*[r...@tel ~]# amportal restart*
*
*
*STOPPING ASTERISK*
*Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl
exist?)*
*Asterisk Stopped*
*
*
*STOPPING FOP SERVER*
*SETTING FILE PERMISSIONS*
*chown: cannot access `/dev/tty9': No such file or directory*
*Permissions OK*
*
*
*STARTING ASTERISK*
*Cannot find specified TTY (9)*
*safe_asterisk: no process killed*
*mpg123: no process killed*
*
*
*-*
*Asterisk could not start!*
*Use 'tail /var/log/asterisk/full' to find out why.*
*-*
*[r...@tel ~]#*
*[r...@tel ~]#*
*[r...@tel ~]# asterisk -g*
*[r...@tel ~]# amportal start*
*
*
*
*
*SETTING FILE PERMISSIONS*
*chown: cannot access `/dev/tty9': No such file or directory*
*Permissions OK*
*
*
*STARTING ASTERISK*
*Asterisk is already running*
*
*
*STARTING FOP SERVER*
*FOP Server Started*

I did a tail and here it is:

*[r...@tel ~]# tail /var/log/asterisk/full*
*[Jun  7 22:17:36] WARNING[4384] chan_dahdi.c: Ignoring any changes to
'userbase' (on reload) at line 23.*
*[Jun  7 22:17:36] WARNING[4384] chan_dahdi.c: Ignoring any changes to
'vmsecret' (on reload) at line 31.*
*[Jun  7 22:17:36] WARNING[4384] chan_dahdi.c: Ignoring any changes to
'hassip' (on reload) at line 35.*
*[Jun  7 22:17:36] WARNING[4384] chan_dahdi.c: Ignoring any changes to
'hasiax' (on reload) at line 39.*
*[Jun  7 22:17:36] WARNING[4384] chan_dahdi.c: Ignoring any changes to
'hasmanager' (on reload) at line 47.*
*[Jun  7 22:17:36] NOTICE[4384] chan_skinny.c: Configuring skinny from
skinny.conf*
*[Jun  7 22:17:36] WARNING[4384] res_musiconhold.c: Cannot open dir
/var/lib/asterisk/moh or dir does not exist*
*[Jun  7 22:17:36] WARNING[4384] res_musiconhold.c: Cannot open dir
/var/lib/asterisk/moh/.nomusic_reserved or dir does not exist*
*[Jun  7 22:17:36] WARNING[4384] res_musiconhold.c: No music on hold classes
configured, disabling music on hold.*
*[Jun  7 22:18:40] ERROR[4479] pbx.c: Did not remove this priority label
(57/vmxopts) from the peer_label_table of context macro-vm, extension vmx!*


Thanks,
Bruce

On Mon, Jun 7, 2010 at 3:29 PM, Steve Edwards wrote:

> On Mon, 7 Jun 2010, bruce bruce wrote:
>
> > CentOS 5.4 and asterisk does stay running after it's loaded by asterisk
> > -g. But the chkconfig --add asterisk doesn't work :(
>
> What does "chkconfig --list asterisk" show?
>
> The "add" command looks in the asterisk script for a line that looks like:
>
># chkconfig: 2345 98 98
>
> This says that chkconfig should create the appropriate links in the
> /etc/rc{x}.d/ hierarchy so that Asterisk will be started at runlevels 2,
> 3, 4, 5 with a start priority of 98 (see "man chkconfig" for details) and
> a stop priority of 98. Since CentOS servers should be running at runlevel
> 3, the 2, 4, and 5 are superfluous.
>
> If there is no such line, chkconfig will not create the appropriate links.
>
> Also, if /etc/init.d/asterisk does not have execute privileges, it will
> not be executed on startup and Asterisk will not be running as expected.
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
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> _
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Re: [asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk

[bruce bruce]

CentOS 5.4 and asterisk does stay running after it's loaded by asterisk -g.
But the chkconfig --add asterisk doesn't work :(

I need FOP and this error should go away as it's annoying. I don't see this
on Trixbox, piaf, or Elastix. It shouldn't be on my install either.

Thanks for the input.

On Sun, Jun 6, 2010 at 11:43 PM, Seann Clark wrote:

> The op_server.pl is part of the Flash Operators Panel, which isn't really
> important to the operation of the PBX, it is just a nice pretty interface
> showing what lines and what groups are active. What O/S are you using? Are
> there any errors in the asterisk logs? Does asterisk stay running after it
> starts?
>
> ~Seann
>
> On 6/6/2010 5:00 PM, bruce bruce wrote:
>
>> Reboot like 10 times and the problem still presists.
>>
>> Also, upon reboot despite having done "chkconfig --add asterisk" asterisk
>> still doesn't load automatically. And amportal start fails. So, I have to do
>> "asterisk -g" first and then amportal start. Wondering if that might be
>> related?
>>
>> Thanks for the input.
>>
>> On Sun, Jun 6, 2010 at 4:47 PM, dotnetdub > dotnet...@gmail.com>> wrote:
>>
>>
>>
>>On 6 June 2010 19:48, bruce bruce ><mailto:bruceb...@gmail.com>> wrote:
>>
>>Hi Guys,
>>
>>Just did an Asterisk 1.6.x (repo install) and FreePBX (source
>>install). When trying to dial a number, I get this:
>>
>>tel*CLI> Use of uninitialized value in hash element at
>>/var/www/html/panel/op_server.pl <http://op_server.pl> line 3367.
>>
>>Use of uninitialized value in concatenation (.) or string at
>>/var/www/html/panel/op_server.pl <http://op_server.pl> line 3372.
>>
>>Use of uninitialized value in pattern match (m//) at
>>/var/www/html/panel/op_server.pl <http://op_server.pl> line 3374.
>>
>>
>>
>>What could be causing that? I searched google and no useful
>>information.
>>
>>Thanks,
>>Bruce
>>
>>
>>Reboot and should go away
>>
>>
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>>
>>
>
>
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Re: [asterisk-users] problem with port 5090 registration

[bruce bruce]

Thanks for the input but it has nothing to do with the trunk configuration
as EXACT same configuration works on another server with iptables disabled.
I disabled iptables on this server as well but it doesn't work.

sip show registery shows a Request Sent.

-Bruce

On Sun, Jun 6, 2010 at 4:58 PM, Tilghman Lesher  wrote:

> On Sunday 06 June 2010 13:46:49 bruce bruce wrote:
> > I have tried every single rule I could into iptables but I can't register
> > this VPS to a provider Spikko. Finally I did an iptable accept on INPUT,
> > OUTPUT, and FORWARD, for ports 0:65000 just to test things and still I
> > can't register to the provider.
> >
> > I can easily register to another provider which uses port 5060 and I can
> > make calls. But sip show registry only show Request Sent for registry and
> > there are 6 tries of Invite and then call fails.
>
> Show us your register line and sip peer configuration (with the username
> and
> password X'ed out).  I suspect a lack of a port configuration somewhere.
>
> --
> Tilghman Lesher
> Digium, Inc. | Senior Software Developer
> twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
> Check us out at: www.digium.com & www.asterisk.org
>
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Re: [asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk

[bruce bruce]

Reboot like 10 times and the problem still presists.

Also, upon reboot despite having done "chkconfig --add asterisk" asterisk
still doesn't load automatically. And amportal start fails. So, I have to do
"asterisk -g" first and then amportal start. Wondering if that might be
related?

Thanks for the input.

On Sun, Jun 6, 2010 at 4:47 PM, dotnetdub  wrote:

>
>
> On 6 June 2010 19:48, bruce bruce  wrote:
>
>> Hi Guys,
>>
>> Just did an Asterisk 1.6.x (repo install) and FreePBX (source install).
>> When trying to dial a number, I get this:
>>
>> tel*CLI> Use of uninitialized value in hash element at
>> /var/www/html/panel/op_server.pl line 3367.
>> Use of uninitialized value in concatenation (.) or string at
>> /var/www/html/panel/op_server.pl line 3372.
>> Use of uninitialized value in pattern match (m//) at /var/www/html/panel/
>> op_server.pl line 3374.
>>
>>
>> What could be causing that? I searched google and no useful information.
>>
>> Thanks,
>> Bruce
>>
>
> Reboot and should go away
>
>
> --
> _
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[asterisk-users] problem with port 5090 registration

[bruce bruce]

Hi Guys,

I have tried every single rule I could into iptables but I can't register
this VPS to a provider Spikko. Finally I did an iptable accept on INPUT,
OUTPUT, and FORWARD, for ports 0:65000 just to test things and still I can't
register to the provider.

I can easily register to another provider which uses port 5060 and I can
make calls. But sip show registry only show Request Sent for registry and
there are 6 tries of Invite and then call fails.

Following is the iptables info:

[r...@tel ~]# service iptables status
Table: nat
Chain PREROUTING (policy ACCEPT)
num  target prot opt source   destination

Chain POSTROUTING (policy ACCEPT)
num  target prot opt source   destination

Chain OUTPUT (policy ACCEPT)
num  target prot opt source   destination

Table: mangle
Chain PREROUTING (policy ACCEPT)
num  target prot opt source   destination

Chain INPUT (policy ACCEPT)
num  target prot opt source   destination

Chain FORWARD (policy ACCEPT)
num  target prot opt source   destination

Chain OUTPUT (policy ACCEPT)
num  target prot opt source   destination

Chain POSTROUTING (policy ACCEPT)
num  target prot opt source   destination

Table: filter
Chain INPUT (policy ACCEPT)
num  target prot opt source   destination
1ACCEPT tcp  --  0.0.0.0/00.0.0.0/0   tcp
dpts:0:65000
2ACCEPT udp  --  0.0.0.0/00.0.0.0/0   udp
dpts:0:65000
3ACCEPT all  --  0.0.0.0/00.0.0.0/0   state
RELATED,ESTABLISHED
4ACCEPT all  --  0.0.0.0/00.0.0.0/0
5ACCEPT icmp --  0.0.0.0/00.0.0.0/0   icmp type
8

Chain FORWARD (policy ACCEPT)
num  target prot opt source   destination
1ACCEPT udp  --  0.0.0.0/00.0.0.0/0   udp
dpts:0:65000
2ACCEPT tcp  --  0.0.0.0/00.0.0.0/0   tcp
dpts:0:65000
3ACCEPT all  --  0.0.0.0/00.0.0.0/0   state
RELATED,ESTABLISHED

Chain OUTPUT (policy ACCEPT)
num  target prot opt source   destination
1ACCEPT udp  --  0.0.0.0/00.0.0.0/0   udp
dpts:0:65000
2ACCEPT tcp  --  0.0.0.0/00.0.0.0/0   tcp
dpts:0:65000
3ACCEPT all  --  0.0.0.0/00.0.0.0/0   state
NEW,RELATED,ESTABLISHED


Thanks,
Bruce
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[asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk

[bruce bruce]

Hi Guys,

Just did an Asterisk 1.6.x (repo install) and FreePBX (source install). When
trying to dial a number, I get this:

tel*CLI> Use of uninitialized value in hash element at /var/www/html/panel/
op_server.pl line 3367.
Use of uninitialized value in concatenation (.) or string at
/var/www/html/panel/op_server.pl line 3372.
Use of uninitialized value in pattern match (m//) at /var/www/html/panel/
op_server.pl line 3374.


What could be causing that? I searched google and no useful information.

Thanks,
Bruce
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Re: [asterisk-users] Best way to limit outgoing calls per trunk

[bruce bruce]

Hi Vardan,

I am using use_dnid=yes and then setting the Account Code in Asterisk
dialplan before sending the call to A2Billing _x. context which
automatically dials. So, before the call goes to A2Billing, I can check to
see if there is a channel up or not. I am not sure how the local channel you
mentioned works. Would appreciate it if you share.

Can you determine the number of channels in the queue?

One of my trunks allows for 3 calls certain time of the day and sometime it
allows for only 1 channel. Hence the need for this.

Thanks,


On Mon, May 31, 2010 at 11:39 AM, Vardan Harutyunyan wrote:

> No, if You use call-limit the call will be dropped.
> How you put your customer on hold?
> If you use queue and the customer hear the music onhold, he will be
> billed for this connection
> I have try use queue and a2b, and I have do all connection using local
> channel, so I have become all is works, and the customer after speaking
> with agents and transferred to international number, is billed only for
> international call.
>
> Sorry for my english, if any question, please write. I will try to explain.
>
> Thanks
>
> --
> Vardan Harutyunyan,
> Senior System Administrator
>
> Enterprise Incubator Foundation
> 123 Hovsep Emin Street,
> Yerevan 0051, Republic of Armenia
> Tel: + 374 10 219735
> Fax: + 374 10 219777
> E-mail: i...@eif.am
> www.eif-it.com
>
> bruce bruce wrote:
> > Thanks for the advice, but I have to keep the customer on hold till the
> > line becomes available. Is that possible by the method you mentioned? I
> > am using A2B 1.7 and Asterisk 1.4.
> >
> > Thanks,
> >
> >
> > On Mon, May 31, 2010 at 2:27 AM, Vardan Harutyunyan  > <mailto:hvarda...@gmail.com>> wrote:
> >
> > Hello,
> >
> > What version of Asterisk You are use?
> > And what version of A2Billing You are use?
> > If You use version 1.4.X of Asterisk You can put call-limit string in
> > sip.conf for this trunk
> >
> > If You use A2B ver 1.7 and Asterk 1.4 you can announce this trunk
> using
> > sip config in A2B, and the are call-limit via web.
> >
> > And how I know, in 1.6 is no more call-limit in sip.conf
> >
> >
> > --
> > Vardan Harutyunyan,
> > Senior System Administrator
> >
> > Enterprise Incubator Foundation
> > 123 Hovsep Emin Street,
> > Yerevan 0051, Republic of Armenia
> > Tel: + 374 10 219735
> > Fax: + 374 10 219777
> > E-mail: i...@eif.am <mailto:i...@eif.am>
> > www.eif-it.com <http://www.eif-it.com>
> >
> > bruce bruce wrote:
> >  > Thanks for that. It very well detailed.
> >  >
> >  > I am not sure if I can use GROUP and GROUP_COUNT now that I see
> > how it's
> >  > used. You see, the call is placed by A2Billing so I don't have a
> > control
> >  > over setting GROUP increase and so if there is a call GROUP_COUNT
> > won't
> >  > work.
> >  >
> >  > I might resort back to using "sed" and "awk" to take output of
> "core
> >  > show channels" and check for it's state. I will appreciate some
> > guru of
> >  > "sed" to to give me a true false for a channel up or not using
> > "sed" and
> >  > "core show channels"
> >  >
> >  > Thanks,
> >  > Bruce
> >  >
> >  > On Sun, May 30, 2010 at 1:47 PM, Jonathan Thurman
> >  > mailto:jonat...@thurmantech.com>
> > <mailto:jonat...@thurmantech.com <mailto:jonat...@thurmantech.com>>>
> > wrote:
> >  >
> >  > On Sun, May 30, 2010 at 9:37 AM, bruce bruce
> > mailto:bruceb...@gmail.com>
> >  > <mailto:bruceb...@gmail.com <mailto:bruceb...@gmail.com>>> wrote:
> >  > > Thanks for the tip. I have been checking those two options.
> Would
> >  > you be
> >  > > able to provide an example of how GROUP or GROUP_COUNT may check
> >  > for a trunk
> >  > > usuage?
> >  >
> >  > Here is how I do it.  It is based on Asterisk 1.6.1.x, and I
> > created a
> >  > generic sub-routine to call for limiting trunks to a specific
> > number
> >  > of calls.  The code is documented, so it should give you a
> > good idea
> >  > of how to use it.
> >  >
&

Re: [asterisk-users] Best way to limit outgoing calls per trunk

[bruce bruce]

Thanks for the advice, but I have to keep the customer on hold till the line
becomes available. Is that possible by the method you mentioned? I am using
A2B 1.7 and Asterisk 1.4.

Thanks,


On Mon, May 31, 2010 at 2:27 AM, Vardan Harutyunyan wrote:

> Hello,
>
> What version of Asterisk You are use?
> And what version of A2Billing You are use?
> If You use version 1.4.X of Asterisk You can put call-limit string in
> sip.conf for this trunk
>
> If You use A2B ver 1.7 and Asterk 1.4 you can announce this trunk using
> sip config in A2B, and the are call-limit via web.
>
> And how I know, in 1.6 is no more call-limit in sip.conf
>
>
> --
> Vardan Harutyunyan,
> Senior System Administrator
>
> Enterprise Incubator Foundation
> 123 Hovsep Emin Street,
> Yerevan 0051, Republic of Armenia
> Tel: + 374 10 219735
> Fax: + 374 10 219777
> E-mail: i...@eif.am
> www.eif-it.com
>
> bruce bruce wrote:
> > Thanks for that. It very well detailed.
> >
> > I am not sure if I can use GROUP and GROUP_COUNT now that I see how it's
> > used. You see, the call is placed by A2Billing so I don't have a control
> > over setting GROUP increase and so if there is a call GROUP_COUNT won't
> > work.
> >
> > I might resort back to using "sed" and "awk" to take output of "core
> > show channels" and check for it's state. I will appreciate some guru of
> > "sed" to to give me a true false for a channel up or not using "sed" and
> > "core show channels"
> >
> > Thanks,
> > Bruce
> >
> > On Sun, May 30, 2010 at 1:47 PM, Jonathan Thurman
> > mailto:jonat...@thurmantech.com>> wrote:
> >
> > On Sun, May 30, 2010 at 9:37 AM, bruce bruce  > <mailto:bruceb...@gmail.com>> wrote:
> >  > Thanks for the tip. I have been checking those two options. Would
> > you be
> >  > able to provide an example of how GROUP or GROUP_COUNT may check
> > for a trunk
> >  > usuage?
> >
> > Here is how I do it.  It is based on Asterisk 1.6.1.x, and I created
> a
> > generic sub-routine to call for limiting trunks to a specific number
> > of calls.  The code is documented, so it should give you a good idea
> > of how to use it.
> >
> > http://thurmantech.com/node/7
> >
> > -Jonathan
> >
> >
> >  >From what I see is that you have to assing certain routes a group
> >  > and then count the group, but how I do include a trunk in the
> group?
> >  > Thanks
> >  >
> >  > On Sat, May 29, 2010 at 7:07 PM, Steve Edwards  > <http://asterisk.org>@sedwards.com <http://sedwards.com>>
> >  > wrote:
> >  >>
> >  >> On Sat, 29 May 2010, bruce bruce wrote:
> >  >>
> >  >> > I am looking to use System() function along with some bash
> > scripting to
> >  >> > determine if a Trunk is being used during certain time of the
> > day or
> >  >> > not. Here is what I have in mind. Please guide me if you know
> > a better
> >  >> > way:
> >  >>
> >  >> Using the GROUP/GROUP_COUNT functions in the dialplan is a
> > better way.
> >  >>
> >  >> Using system() will mean creating a bunch of processes (each
> >  >> sed/awk/cut/etc command).
> >  >>
> >  >> --
> >  >> Thanks in advance,
> >  >>
> >
> -
> >  >> Steve Edwards sedwa...@sedwards.com
> > <mailto:sedwa...@sedwards.com>  Voice: +1-760-468-3867 PST
> >  >> Newline  Fax:
> > +1-760-731-3000
> >  >>
> >  >> --
> >  >>
> > _
> >  >> -- Bandwidth and Colocation Provided by
> > http://www.api-digital.com --
> >  >> New to Asterisk? Join us for a live introductory webinar every
> > Thurs:
> >  >> http://www.asterisk.org/hello
> >  >>
> >  >> asterisk-users mailing list
> >  >> To UNSUBSCRIBE or update options visit:
> >  >> http://lists.digium.com/mailman/listinfo/asterisk-users
> >  >
> >  >
> >   

Re: [asterisk-users] Best way to limit outgoing calls per trunk

[bruce bruce]

Thanks for that. It very well detailed.

I am not sure if I can use GROUP and GROUP_COUNT now that I see how it's
used. You see, the call is placed by A2Billing so I don't have a control
over setting GROUP increase and so if there is a call GROUP_COUNT won't
work.

I might resort back to using "sed" and "awk" to take output of "core show
channels" and check for it's state. I will appreciate some guru of "sed" to
to give me a true false for a channel up or not using "sed" and "core show
channels"

Thanks,
Bruce

On Sun, May 30, 2010 at 1:47 PM, Jonathan Thurman
wrote:

> On Sun, May 30, 2010 at 9:37 AM, bruce bruce  wrote:
> > Thanks for the tip. I have been checking those two options. Would you be
> > able to provide an example of how GROUP or GROUP_COUNT may check for a
> trunk
> > usuage?
>
> Here is how I do it.  It is based on Asterisk 1.6.1.x, and I created a
> generic sub-routine to call for limiting trunks to a specific number
> of calls.  The code is documented, so it should give you a good idea
> of how to use it.
>
> http://thurmantech.com/node/7
>
> -Jonathan
>
>
> >From what I see is that you have to assing certain routes a group
> > and then count the group, but how I do include a trunk in the group?
> > Thanks
> >
> > On Sat, May 29, 2010 at 7:07 PM, Steve Edwards  sedwards.com>
> > wrote:
> >>
> >> On Sat, 29 May 2010, bruce bruce wrote:
> >>
> >> > I am looking to use System() function along with some bash scripting
> to
> >> > determine if a Trunk is being used during certain time of the day or
> >> > not. Here is what I have in mind. Please guide me if you know a better
> >> > way:
> >>
> >> Using the GROUP/GROUP_COUNT functions in the dialplan is a better way.
> >>
> >> Using system() will mean creating a bunch of processes (each
> >> sed/awk/cut/etc command).
> >>
> >> --
> >> Thanks in advance,
> >>
> -
> >> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867
> PST
> >> Newline  Fax:
> +1-760-731-3000
> >>
> >> --
> >> _
> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >> New to Asterisk? Join us for a live introductory webinar every Thurs:
> >>   http://www.asterisk.org/hello
> >>
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> >   http://www.asterisk.org/hello
> >
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> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
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Re: [asterisk-users] Best way to limit outgoing calls per trunk

[bruce bruce]

Thanks for the tip. I have been checking those two options. Would you be
able to provide an example of how GROUP or GROUP_COUNT may check for a trunk
usuage? From what I see is that you have to assing certain routes a group
and then count the group, but how I do include a trunk in the group?

Thanks

On Sat, May 29, 2010 at 7:07 PM, Steve Edwards wrote:

> On Sat, 29 May 2010, bruce bruce wrote:
>
> > I am looking to use System() function along with some bash scripting to
> > determine if a Trunk is being used during certain time of the day or
> > not. Here is what I have in mind. Please guide me if you know a better
> > way:
>
> Using the GROUP/GROUP_COUNT functions in the dialplan is a better way.
>
> Using system() will mean creating a bunch of processes (each
> sed/awk/cut/etc command).
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Best way to limit outgoing calls per trunk

[bruce bruce]

Thanks for the input.

Is there no Asterisk command in dialplan to return true or false or number
of channels per trunk rather than going to bash script?

And if I do the bash script how can I use "sed" and "awk" to search for the
right words to see if trunk is in use or not.

A bit detail would be really helpful.

Thanks,
Bruce

On Sat, May 29, 2010 at 5:28 PM, Zeeshan Zakaria  wrote:

> Should be solid. After all munin also works on the same lines and it works
> solid.
>
> Zeeshan A Zakaria
>
> --
> Sent from my Android phone with K-9 Mail.
>
> On 2010-05-29 5:12 PM, "bruce bruce"  wrote:
>
> Hi Guys,
>
> I am looking to use System() function along with some bash scripting to
> determine if a Trunk is being used during certain time of the day or not.
> Here is what I have in mind. Please guide me if you know a better way:
>
> exten => s,1,answer
> exten => s,n,System(/tmp/check.sh)
>
> check.sh:
> check EPOCH time => do an IF for certain times => Allow mutiple calls in
> certain times and only single call at certain times
> return back to Asterisk context and report if Trunk would allow more
> channels or not...
>
> Something along those lines. Should this be a solid thing to do? I am
> looking to use GotoIF and `asterisk -rx "sip show channels"` to grab results
> or `asterisk -rx "core show channels"`
>
> Thanks
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
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>
>
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[asterisk-users] Best way to limit outgoing calls per trunk

[bruce bruce]

Hi Guys,

I am looking to use System() function along with some bash scripting to
determine if a Trunk is being used during certain time of the day or not.
Here is what I have in mind. Please guide me if you know a better way:

exten => s,1,answer
exten => s,n,System(/tmp/check.sh)

check.sh:
check EPOCH time => do an IF for certain times => Allow mutiple calls in
certain times and only single call at certain times
return back to Asterisk context and report if Trunk would allow more
channels or not...

Something along those lines. Should this be a solid thing to do? I am
looking to use GotoIF and `asterisk -rx "sip show channels"` to grab results
or `asterisk -rx "core show channels"`

Thanks
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Re: [asterisk-users] "pri show version" still shows old version despite doing a make && make clean && make install for v1.4.11

[bruce bruce]

Hi Guys,

Anyone else can comment on this or give me their thoughts please? I just
want to know if someone can confirm the output for "make install" in new
LibPRI directory.

Thanks,
Bruce


On Fri, May 28, 2010 at 12:58 PM, bruce bruce  wrote:

> Thanks for the input. Yes, I did do a restart of Asterisk and the system
> but changes do no show. Is it normal to not see a "Successful" message after
> doing "make install" within the new LibPri library?
>
> Thanks,
> Bruce
>
> On Thu, May 27, 2010 at 9:41 PM, Tim Nelson  wrote:
>
>> - "bruce bruce"  wrote:
>> >What am I doing wrong that it's not update to 1.4.11?
>> >Thanks, Bruce
>> --
>>
>> Did you restart your services to ensure the new library was picked up?
>>
>> --Tim
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Re: [asterisk-users] "pri show version" still shows old version despite doing a make && make clean && make install for v1.4.11

[bruce bruce]

Thanks for the input. Yes, I did do a restart of Asterisk and the system but
changes do no show. Is it normal to not see a "Successful" message after
doing "make install" within the new LibPri library?

Thanks,
Bruce

On Thu, May 27, 2010 at 9:41 PM, Tim Nelson  wrote:

> - "bruce bruce"  wrote:
> >What am I doing wrong that it's not update to 1.4.11?
> >Thanks, Bruce
> --
>
> Did you restart your services to ensure the new library was picked up?
>
> --Tim
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
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[asterisk-users] "pri show version" still shows old version despite doing a make && make clean && make install for v1.4.11

[bruce bruce]

Hi Guys,

I am running a PBXinaFLASH server. I replaced contents of /usr/src/libpri
with the new version of Libpri v1.4.11. The installed one was v1.4.10.
System is running Asterisk  1.4.21.2.

I did the following after:

cd /usr/src/libpri/
make
make clean
make install

Install end with these lines.:

*ln -sf libpri.so.1.4 libpri.so*
*mkdir -p /usr/lib*
*mkdir -p /usr/include*
*install -m 644 libpri.h /usr/include*
*install -m 755 libpri.so.1.4 /usr/lib*
*#if [ -x /usr/sbin/sestatus ] && ( /usr/sbin/sestatus | grep "SELinux
status:" | grep -q "enabled"); then /sbin/restorecon -v
/usr/lib/libpri.so.1.4; fi*
*( cd /usr/lib ; ln -sf libpri.so.1.4 libpri.so)*
*install -m 644 libpri.a /usr/lib*
*if test $(id -u) = 0; then /sbin/ldconfig -n /usr/lib; fi*


Is this ^ installed properly? Don't I get a successful message?
And here is the result:
*r...@pbx:/usr/src/libpri $ asterisk -rx "pri show version"*
*libpri version: 1.4.10.2*

What am I doing wrong that it's not update to 1.4.11?

Thanks,
Bruce
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Re: [asterisk-users] q931.c modifications for CLID Presentation

[bruce bruce]

Anyone can comment on this please?

Is it right to assume that if you own a PRI Caller ID always comes through
even if customer used *67 feature to block their CLID?
I understand that is true of calling a Toll-Free number.

Does Asterisk or LibPRI somewhere in the code abide by some standard to
strip or hide the CLID if Callee requested private presentation?

Thanks

On Sat, May 15, 2010 at 4:14 PM, bruce bruce  wrote:

> Hi Guys,
>
> We have a problem with Caller ID not being displayed. I want to test
> everything to see where the problem is with the incoming Caller ID and why
> it's not displaying.
>
> I am tracking this down to "Presentation prohibited of network provided
> number" even though the Caller doesn't use *67 and even though they haven't
> asked their provider to block their CLID for outbound.
>
> I want to make a modification to q931.c or pri_facility (whichever
> responsible) to ignore the request from the network to prohibit CLID and to
> show it so that I can find out exactly where the problem lies. Can you
> please tell me which "if" is related to that in q931.c or pri_facility.c?
>
> Thanks
>
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Re: [asterisk-users] Libpri 1.4.11 Released

[bruce bruce]

Thanks for the update. How to upgrade to the latest stable release without
compliling Asterisk again? Can you please explain and detail the commands?
We are running PBXinaFlash with LibPRI 1.4.10.1 which gives lots of
problems.

Thanks

On Wed, May 26, 2010 at 12:27 PM, Asterisk Development Team <
asteriskt...@digium.com> wrote:

> The Asterisk Development Team has announced the release of version
> 1.4.11 of libpri. This release is available for immediate download at
> http://downloads.asterisk.org/pub/telephony/libpri/
>
> This release contains many fixes and new features, among them being:
>
> 1.) Support for NT-PTMP BRI links, including support for multiple TEIs
> and connecting of BRI phones.
>
> 2.) Support for allowing persistent Q.921 drops on both NT and TE PTMP
> links, as well as automatically requesting that Q.921 data links
> reactivate when needed by Q.931.
>
> 3.) T309 is enabled by default.
>
> 4.) Problems with Keypad Facility Digits were addressed.
>
> 5.) A number of additional service related features were added:
> Connected Line Information, HOLD/RELEASE support, Call Deflection/Call
> Rerouting, as well as partial subaddress support.  They are supported in
> the Q.SIG and EuroISDN switch types, and most currently require using
> the trunk version of Asterisk.
>
> 6.) Many potential and realized Q.921 related problems, particularly
> during retransmissions and other scenarios involving medium to high
> packet loss.
>
> For a full list of changes in the current release candidates, please see
> the ChangeLog:
>
>
> http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1.4.11
>
> Thank you for your continued support of Asterisk!
>
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Re: [asterisk-users] Softphones on thin clients...

[bruce bruce]

Is the Java soft phone an open source or obtainable? I am just checking
their site and it seems they only provide service??!!

Their java web based client is built neatly. Would like to test that on my
servers.

On Thu, May 20, 2010 at 3:21 PM,  wrote:

> I've used HP Thin Clients as embedded hosts for Asterisk. The T5700
> models that I have are 1 GHz CPUs, more recent models should be able to
> run a soft phone without too much trouble. They all have local USB
> ports, making USB headsets as good solution.
>
> Another alternative might be to used a soft phone implemented as a web
> plug-in or activex object. Tim Panton of PhoneFromHere.com has a great
> Java soft phone object that we use to make G.722 calls to the ZipDX
> conference bridge for the VoIP Users Conference every week.
>
> Michael Graves
> mgraves  mstvp.com
> o(713) 861-4005
> c(713) 201-1262
> sip:mjgra...@mstvp.onsip.com 
> skype mjgraves
>
> >  Original Message 
> > Subject: Re: [asterisk-users] Softphones on thin clients...
> > From: Carlos Chavez 
> > Date: Thu, May 20, 2010 1:36 pm
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > 
> >
> >
> > > -Original Message-
> > > From: asterisk-users-boun...@lists.digium.com
> > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
> Howes
> > > Sent: Thursday, May 20, 2010 1:51 PM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: Re: [asterisk-users] Softphones on thin clients...
> > >
> > >
> > > On 20 May 2010, at 18:35, Carlos Chavez wrote:
> > > >   I am worried about conflicts when running 10 softphones on the same
> > > > server since they will all try to use por 5060.
> > >
> > > And the fact most terminal services servers/clients still don't support
> > > audio input.. only output..
> >
> >   Since the little box has a MIC jack I suppose that it should
> support
> > audio input.  These boxes will be running Windows and using Eyebeam.
> >
> > --
> > Telecomunicaciones Abiertas de México S.A. de C.V.
> > Carlos Chávez Prats
> > Director de Tecnología
> > +52-55-91169161 ext 2001--
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> >
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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Re: [asterisk-users] Asterisk and RFC 3261

[bruce bruce]

That is the RFC number for SIP. Yes, Asterisk is compliant with RFC. I am
not sure to what degree but I haven't ever faced non-compliance on SIP RFC 3261
ever with any provider.

-Bruce

On Wed, May 19, 2010 at 2:28 PM, Tarek Sawah  wrote:

>
>
> Greetings List,Trying to interconnect with a new provider.. the require
> a compliance with RFC 3261  so knowing less than needed about RFC
> documentations.. i would like to know if Asterisk is actually in compliance
> with RFC 3261 or not.. Can any one help with this?
> Regards
> --
> Tarek Sawah
> Integrated Digital Systems
> CCNA, MCSE, RHCE, VoIP
> USA: +1 347 562 2308
>
>
>
>
>
> _
> The New Busy is not the old busy. Search, chat and e-mail from your inbox.
>
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Re: [asterisk-users] Commands to upgrade to latest Libpri? can I upgrade without touching Asterisk?

[bruce bruce]

When I do "show pri version" I still see 1.4.10. Is that right or should I
see 1.4.10.2 since I upgraded it.

I did the install but shouldn't I get an "Install sucesful message" ?

*Just did a "make clean && make && make install" and output is:*
gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC   -MD
-MT copy_string.o -MF .copy_string.o.d -MP -c -o copy_string.o copy_string.c
gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC   -MD
-MT pri.o -MF .pri.o.d -MP -c -o pri.o pri.c
gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC   -MD
-MT q921.o -MF .q921.o.d -MP -c -o q921.o q921.c
gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC   -MD
-MT prisched.o -MF .prisched.o.d -MP -c -o prisched.o prisched.c
gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC   -MD
-MT q931.o -MF .q931.o.d -MP -c -o q931.o q931.c
gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC   -MD
-MT pri_facility.o -MF .pri_facility.o.d -MP -c -o pri_facility.o
pri_facility.c
gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC   -MD
-MT version.o -MF .version.o.d -MP -c -o version.o version.c
ar rcs libpri.a copy_string.o pri.o q921.o prisched.o q931.o pri_facility.o
version.o
ranlib libpri.a
gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC   -MD
-MT copy_string.lo -MF .copy_string.lo.d -MP -c -o copy_string.lo
copy_string.c
gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC   -MD
-MT pri.lo -MF .pri.lo.d -MP -c -o pri.lo pri.c
gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC   -MD
-MT q921.lo -MF .q921.lo.d -MP -c -o q921.lo q921.c
gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC   -MD
-MT prisched.lo -MF .prisched.lo.d -MP -c -o prisched.lo prisched.c
gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC   -MD
-MT q931.lo -MF .q931.lo.d -MP -c -o q931.lo q931.c
gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC   -MD
-MT pri_facility.lo -MF .pri_facility.lo.d -MP -c -o pri_facility.lo
pri_facility.c
gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC   -MD
-MT version.lo -MF .version.lo.d -MP -c -o version.lo version.c
gcc -shared -Wl,-hlibpri.so.1.4 -o libpri.so.1.4 copy_string.lo pri.lo
q921.lo prisched.lo q931.lo pri_facility.lo version.lo
/sbin/ldconfig -n .
ln -sf libpri.so.1.4 libpri.so
mkdir -p /usr/lib
mkdir -p /usr/include
install -m 644 libpri.h /usr/include
install -m 755 libpri.so.1.4 /usr/lib
if [ -x /usr/sbin/sestatus ] && ( /usr/sbin/sestatus | grep "SELinux
status:" | grep -q "enabled"); then /sbin/restorecon -v
/usr/lib/libpri.so.1.4; fi
( cd /usr/lib ; ln -sf libpri.so.1.4 libpri.so)
install -m 644 libpri.a /usr/lib
if test $(id -u) = 0; then /sbin/ldconfig -n /usr/lib; fi

Thanks,
Bruce



On Mon, May 17, 2010 at 4:03 PM, bruce bruce  wrote:

> Thanks for the help Tzafrir.
>
> I think for libpri you meant >= 1.4.x rather than 1.4.4 as the latest
> version available is 1.4.10.2or maybe 1.4.10 is greater than 1.4.4 ?!
> Why haven't they changed the name to 1.5.0. I never get the nomenclature for
> these things.
>
> Thanks again,
> Bruce
>
>
> On Mon, May 17, 2010 at 3:48 PM, Tzafrir Cohen 
> wrote:
>
>> On Mon, May 17, 2010 at 03:22:04PM -0400, bruce bruce wrote:
>> > Hi Guys,
>> >
>> > I have to upgrade to latest Libpri 1.4.10.2 due to an existing bug in
>> the
>> > current 1.4.10 version. I am running Asterisk 1.4.x (in fact it is a
>> > PBXinaFLASH system).
>> >
>> > How can I upgrade to the latest Libpri? Do I need to re-install
>> Asterisk?
>> > Won't that break the box?
>>
>> If you upgrade from a version of libpri that is >= 1.4.4 , you just need
>> to rebuild and reinstall libpri.
>>
>>
>> >
>> > Can I simply do this to upgrade:
>> >
>> > *rm /usr/src/libpri/*.**
>> >
>> > *rm -rf /usr/src/libpri/**
>>
>> Not really needed
>>
>> >
>> >
>> > Download the new libpri and put files in folder /usr/src/libpri/
>> >
>> > *cd /usr/src/libpri && make clean && make && make install*
>>
>> Basically, yes.
>>
>> 'make clean' is not streactly needed, though it is harmless.
>>
>> --
>>   Tzafrir Cohen
>> icq#16849755  
>> jabber:tzafrir.co...@xorcom.com
>> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
>> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
>>
>> --
>> _
>> -- Bandwidth and

Re: [asterisk-users] Commands to upgrade to latest Libpri? can I upgrade without touching Asterisk?

[bruce bruce]

Thanks for the help Tzafrir.

I think for libpri you meant >= 1.4.x rather than 1.4.4 as the latest
version available is 1.4.10.2or maybe 1.4.10 is greater than 1.4.4 ?!
Why haven't they changed the name to 1.5.0. I never get the nomenclature for
these things.

Thanks again,
Bruce

On Mon, May 17, 2010 at 3:48 PM, Tzafrir Cohen wrote:

> On Mon, May 17, 2010 at 03:22:04PM -0400, bruce bruce wrote:
> > Hi Guys,
> >
> > I have to upgrade to latest Libpri 1.4.10.2 due to an existing bug in the
> > current 1.4.10 version. I am running Asterisk 1.4.x (in fact it is a
> > PBXinaFLASH system).
> >
> > How can I upgrade to the latest Libpri? Do I need to re-install Asterisk?
> > Won't that break the box?
>
> If you upgrade from a version of libpri that is >= 1.4.4 , you just need
> to rebuild and reinstall libpri.
>
>
> >
> > Can I simply do this to upgrade:
> >
> > *rm /usr/src/libpri/*.**
> >
> > *rm -rf /usr/src/libpri/**
>
> Not really needed
>
> >
> >
> > Download the new libpri and put files in folder /usr/src/libpri/
> >
> > *cd /usr/src/libpri && make clean && make && make install*
>
> Basically, yes.
>
> 'make clean' is not streactly needed, though it is harmless.
>
> --
>   Tzafrir Cohen
> icq#16849755  
> jabber:tzafrir.co...@xorcom.com
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
>
> --
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[asterisk-users] Commands to upgrade to latest Libpri? can I upgrade without touching Asterisk?

[bruce bruce]

Hi Guys,

I have to upgrade to latest Libpri 1.4.10.2 due to an existing bug in the
current 1.4.10 version. I am running Asterisk 1.4.x (in fact it is a
PBXinaFLASH system).

How can I upgrade to the latest Libpri? Do I need to re-install Asterisk?
Won't that break the box?

Can I simply do this to upgrade:

*rm /usr/src/libpri/*.**

*rm -rf /usr/src/libpri/**


Download the new libpri and put files in folder /usr/src/libpri/

*cd /usr/src/libpri && make clean && make && make install*



Thanks,
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[asterisk-users] PRI down due to chan_zap.c: No more room in scheduler....Got SABME and Sending Unnumbered Acknowledgement...Any thoughts?

[bruce bruce]

Hi Guys,

Running the following with a Sangoma A101D PRI card:

*Asterisk 1.4.21.2*
*LibPRI version: 1.4.10*

No inbound or outbound calls can be made. In fact Asterisk CLI doesn't show
any activity. Problem goes away on restart of the system or maybe asterisk.
I see post about upgrading Libpri to 1.4.10.2 and then I see posts that even
that didn't work. Anyone can weigh in this please?

tail -f /var/log/asterisk/full
*[2010-05-17 13:45:28] ERROR[8212] chan_zap.c: Asked to delete sched id
-1???*
*[2010-05-17 13:45:28] ERROR[8212] chan_zap.c: No more room in scheduler*


pri debug span 1 (repeats continuously):
*-- Got SABME from network peer.*
*Sending Unnumbered Acknowledgement*


pri intense debug span 1 (repeats continuously):
*< [ 02 01 7f ]*
*pbx*CLI>*
*< Unnumbered frame:*
*< SAPI: 00  C/R: 1 EA: 0*
*<  TEI: 000EA: 1*
*<   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode
extended) ]*
*< 0 bytes of data*
*Handling message for SAPI/TEI=0/0*
*-- Got SABME from network peer.*
*Sending Unnumbered Acknowledgement*
*
*
*> [ 02 01 73 ]*
*
*
*> Unnumbered frame:*
*> SAPI: 00  C/R: 1 EA: 0*
*>  TEI: 000EA: 1*
*>   M3: 3   P/F: 1 M2: 0 11: 3  [ UA (unnumbered acknowledgement) ]*
*> 0 bytes of data*
*-- Restarting T203 timer*



Thanks,
Bruce
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Re: [asterisk-users] OK, I'm stumped

[bruce bruce]

Maybe drop the call in a Meetme room and have an announcement?

On Sun, May 16, 2010 at 10:15 AM, Bruce Ferrell wrote:

> I'm trying to make an AMI call.  I want to call a number, play an
> announcement when the call is answered, then call a second number and
> connect the two when the second call is answered.
>
> I an able to make a simple call to two numbers and connect them using
> the manager API but playing the announcement has me beat.
>
> Suggestions anyone?
>
> Bruce Ferrell
>
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Re: [asterisk-users] Re-compiling q931.c

[bruce bruce]

Thanks for the input.

I need a bit more clarification.

I just changed the condition for an "if" in the q931.c from Equal to Not
Equal. In order to get that reflected do I have to do "gcc q931.c" to
recompile it so Asterisk can read it? or does asterisk read the .c file?

Thanks again,
Bruce

On Sat, May 15, 2010 at 4:56 PM, Tzafrir Cohen wrote:

> On Sat, May 15, 2010 at 04:32:19PM -0400, bruce bruce wrote:
> > Hi Guys,
> >
> > Can q931.c be re-compiled using gcc or something else without the need to
> > re-do the whole libpri? Some changes were made to q931.c and I want those
> to
> > be reflected in .a .o .so .lo files as I think those are the files read
> by
> > Asterisk vs the .c file.
>
> Given that libpri is such a small library, I wouldn't bother. If the
> change did not break the binary ABI (e.g.: changed the size of a struct
> that is exposed through some interface, added/removed variables to some
> function) there should be no issue here.
>
> If you want to test things, try just building libpri (and not installing
> it) and start asterisk with:
>
>  LD_LIBRARY_PATH=/path/to/libpri/source asterisk
>
> --
>   Tzafrir Cohen
> icq#16849755  
> jabber:tzafrir.co...@xorcom.com
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
>
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[asterisk-users] Re-compiling q931.c

[bruce bruce]

Hi Guys,

Can q931.c be re-compiled using gcc or something else without the need to
re-do the whole libpri? Some changes were made to q931.c and I want those to
be reflected in .a .o .so .lo files as I think those are the files read by
Asterisk vs the .c file.

Thanks,
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[asterisk-users] q931.c modifications for CLID Presentation

[bruce bruce]

Hi Guys,

We have a problem with Caller ID not being displayed. I want to test
everything to see where the problem is with the incoming Caller ID and why
it's not displaying.

I am tracking this down to "Presentation prohibited of network provided
number" even though the Caller doesn't use *67 and even though they haven't
asked their provider to block their CLID for outbound.

I want to make a modification to q931.c or pri_facility (whichever
responsible) to ignore the request from the network to prohibit CLID and to
show it so that I can find out exactly where the problem lies. Can you
please tell me which "if" is related to that in q931.c or pri_facility.c?

Thanks
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Re: [asterisk-users] aastra pt 480e phone

[bruce bruce]

Unplugging just turns off the phone and has no effect on the settings. You
can not "damage" the phone by tampering configurations but you can mess up
the settings and it might not register, send, or receive calls.

User manu for your reference:

http://www.aastra.com/cps/rde/xbcr/SID-3D8CCB6A-E9BBA0B7/04/480e_ma_en_0306.pdf

Overall instructions:
http://www.aastra.com/cps/rde/xchg/SID-3D8CCB73-FE677A10/04/hs.xsl/19493.htm#dl_instructions

I
couldn't find how to access Web Configurator but I guess it would be the IP
address. And if username/password on this is the same as the newer phones
from Aastra then it would be admin/2

-Bruce

On Fri, May 14, 2010 at 12:38 AM, michael capelle <
michael.cape...@charter.net> wrote:

> hello
> i hope i am posting to the right list, i am a totally blind user, and i
> want
> to reprogram my aastra pt 480e phone, my friend used the web configurator,
> but i think he programmed thw wrong codes, a few questions, is it possible
> to damage the phone by programming it wrong? also, how does one reset it
> literally to factory defaults? as unplugging it and replugging it back into
> the wall didn't work.
> regards
> a happy aastra user.
>
>
> --
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[asterisk-users] Do you think my server is being attacked?

[bruce bruce]

Hello Everyone,

Are these indications of attacks on this system? I specifically have port 22
disabled at all times and only port forward it to server when I access SSH
for a minute or so. Shouldn't UNKNOWN be an actual IP address?

*/var/log/secure:*

May 14 00:35:39 pbx sshd[9011]: Did not receive identification string from
UNKNOWN
May 14 00:36:09 pbx sshd[9040]: Did not receive identification string from
UNKNOWN
May 14 00:36:39 pbx sshd[9075]: Did not receive identification string from
UNKNOWN
May 14 00:37:10 pbx sshd[9102]: Did not receive identification string from
UNKNOWN
May 14 00:37:40 pbx sshd[9139]: Did not receive identification string from
UNKNOWN
May 14 00:38:11 pbx sshd[9166]: Did not receive identification string from
UNKNOWN
May 14 00:38:41 pbx sshd[9195]: Did not receive identification string from
UNKNOWN
May 14 00:39:11 pbx sshd[9230]: Did not receive identification string from
UNKNOWN
May 14 00:39:42 pbx sshd[9250]: Did not receive identification string from
UNKNOWN
May 14 00:40:13 pbx sshd[9294]: Did not receive identification string from
UNKNOWN
May 14 00:40:44 pbx sshd[9329]: Did not receive identification string from
UNKNOWN
May 14 00:41:14 pbx sshd[9366]: Did not receive identification string from
UNKNOWN
May 14 00:41:44 pbx sshd[9401]: Did not receive identification string from
UNKNOWN
May 14 00:42:18 pbx sshd[9437]: Did not receive identification string from
UNKNOWN
May 14 00:42:48 pbx sshd[9457]: Did not receive identification string from
UNKNOWN
May 14 00:43:19 pbx sshd[9492]: Did not receive identification string from
UNKNOWN
May 14 00:43:49 pbx sshd[9521]: Did not receive identification string from
UNKNOWN
May 14 00:44:20 pbx sshd[9564]: Did not receive identification string from
UNKNOWN
May 14 00:44:50 pbx sshd[9600]: Did not receive identification string from
UNKNOWN
May 14 00:45:20 pbx sshd[9636]: Did not receive identification string from
UNKNOWN
May 14 00:45:51 pbx sshd[9663]: Did not receive identification string from
UNKNOWN
May 14 00:46:21 pbx sshd[9692]: Did not receive identification string from
UNKNOWN
May 14 00:46:51 pbx sshd[9721]: Did not receive identification string from
UNKNOWN
May 14 00:47:21 pbx sshd[9756]: Did not receive identification string from
UNKNOWN
May 14 00:47:52 pbx sshd[9792]: Did not receive identification string from
UNKNOWN

Thanks,
Bruce
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[asterisk-users] Sangoma A101D PRI failing with ERROR - -- Got SABME from network peer. Sending Unnumbered Acknowledgement

[bruce bruce]

Hi Guys,

Anyone might know why this error keeps showing up and inbound/outbound is
not working on a Bell PRI with Sangoma A101D?

-- Got SABME from network peer.
Sending Unnumbered Acknowledgement

No calls can be made inbound/outbound.

Keeps repeating. No alarms ON and no changes been made to the system.
Stopped all a sudden. Asterisk CLI doesn't show anything with full verbose
for both inbound and outbound.

pbx*CLI> pri show span 1
Primary D-channel: 24
Status: Provisioned, Up, Active
Switchtype: National ISDN
Type: CPE
Window Length: 0/7
Sentrej: 0
SolicitFbit: 1
Retrans: 0
Busy: 0
Overlap Dial: 0
Logical Channel Mapping: 0
T200 Timer: 1000
T203 Timer: 1
T305 Timer: 3
T308 Timer: 4000
T309 Timer: -1
T313 Timer: 4000
N200 Counter: 3

Thanks,
Bruce
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Re: [asterisk-users] PRI - Native ZAP bridge fails - Is this my patch?

[bruce bruce]

Thanks. Yeah, that was the issue. I was requesting RLT and it wasn't turned
ON with the provider. Your mentioned solution fixed it.

-Bruce

On Fri, Apr 30, 2010 at 9:59 AM, Klaus Darilion <
klaus.mailingli...@pernau.at> wrote:

> The disconnect is RECEIVED by Asterisk. So there is a problem with the
> other party.
>
> You are sending FACILITY - maybe the other party does not like FACILITY and
> hangs up.
>
> IIRC there is a setting in zapata.conf to enable/disable FACILITY.
>
> regards
> klaus
>
> Am 10.04.2010 21:46, schrieb bruce bruce:
>
>  Hi Guys,
>>
>> I am calling out 416-999- on Channel 1 of PRI and then calling
>> 416-999- on Channel 2 of PRI. When the two channels are going to be
>> ZAP native bridged, both channels hangup and CLI show PRI cause (16).
>>
>> Asterisk Verbose *(Channel 1 already connected to party)*:
>> -- Requested transfer capability: 0x00 - SPEECH
>> -- Called g0/416999
>> -- Zap/2-1 is proceeding passing it to Zap/1-1
>> -- Zap/2-1 is ringing
>> -- Zap/2-1 answered Zap/1-1
>> -- Native bridging Zap/1-1 and Zap/2-1
>> -- Channel 0/1, span 1 got hangup request, cause 16
>> -- Hungup 'Zap/2-1'
>>   == Spawn extension (zap-bridge, s, 8) exited non-zero on 'Zap/1-1'
>> -- Hungup 'Zap/1-1'
>>
>> Here is PRI debug, starting just before Channel two is connected until
>> both channels are disconnected *(maybe FACILITY 98 is of interest?!)*:
>>
>> < Message type: CONNECT (7)
>> q931.c:3626 q931_receive: call 32865 on channel 2 enters state 10 (Active)
>>  > Protocol Discriminator: Q.931 (8)  len=5
>>  > Call Ref: len= 2 (reference 97/0x61) (Originator)
>>  > Message type: CONNECT ACKNOWLEDGE (15)
>> -- Zap/2-1 answered Zap/1-1
>> -- Native bridging Zap/1-1 and Zap/2-1
>>  > Protocol Discriminator: Q.931 (8)  len=27
>>  > Call Ref: len= 2 (reference 96/0x60) (Originator)
>>  > Message type: FACILITY (98)
>>  > [1c 14 91 a1 11 02 01 06 06 07 2a 86 48 ce 15 00 08 30 03 02 01 61]
>>  > Facility (len=22, codeset=0) [ 0x91, 0xA1, 0x11, 0x02, 0x01, 0x06,
>> 0x06, 0x07, '*', 0x86, 'H', 0xCE, 0x15, 0x00, 0x08, '0', 0x03, 0x02,
>> 0x01, 'a' ]
>> PROTOCOL 11
>> A1 0011 (CONTEXT SPECIFIC [1])
>>   02 0001 06 (INTEGER: 6)
>>   06 0007 2A 86 48 CE 15 00 08 (OBJECTIDENTIFIER: 2a 86 48 ce 15 00 08)
>>   30 0003 (SEQUENCE)
>> 02 0001 61 (INTEGER: 97)
>> < Protocol Discriminator: Q.931 (8)  len=9
>> < Call Ref: len= 2 (reference 96/0x60) (Terminator)
>> < Message type: DISCONNECT (69)
>> < [08 02 80 90]
>> < Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0
>>  Location: User (0)
>> <  Ext: 1  Cause: Normal Clearing (16), class = Normal
>> Event (1) ]
>> -- Processing IE 8 (cs0, Cause)
>> q931.c:3826 q931_receive: call 32864 on channel 1 enters state 12
>> (Disconnect Indication)
>> -- Channel 0/1, span 1 got hangup request, cause 16
>> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate
>> Connect Request
>> q931.c:3015 q931_disconnect: call 32865 on channel 2 enters state 11
>> (Disconnect Request)
>>  > Protocol Discriminator: Q.931 (8)  len=9
>>  > Call Ref: len= 2 (reference 97/0x61) (Originator)
>>  > Message type: DISCONNECT (69)
>>  > [08 02 81 90]
>>  > Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0
>>  Location: Private network serving the local user (1)
>>  >  Ext: 1  Cause: Normal Clearing (16), class = Normal
>> Event (1) ]
>> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication,
>> peerstate Disconnect Request
>> q931.c:2967 q931_release: call 32864 on channel 1 enters state 19
>> (Release Request)
>>  > Protocol Discriminator: Q.931 (8)  len=9
>>  > Call Ref: len= 2 (reference 96/0x60) (Originator)
>>  > Message type: RELEASE (77)
>>  > [08 02 81 90]
>>  > Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0
>>  Location: Private network serving the local user (1)
>>  >  Ext: 1  Cause: Normal Clearing (16), class = Normal
>> Event (1) ]
>> -- Hungup 'Zap/1-1'
>> < Protocol Discriminator: Q.931 (8)  len=5
>> < Call Ref: len= 2 (reference 96/0x60) (Terminator)
>> < Message type: RELEASE COMPLETE (90)
>> q931.c:3766 q931_receive: call 32864 on channel 1 enters state 0 (Null)
>> NEW_HANGUP DEB

[asterisk-users] How to debug the problem of Asterisk using so much of CPU percentage...?

[bruce bruce]

Hi Everyone,

How is this possible? How can I go about debugging this? I think that the
sound chopping and choking is also related to this. I have never seen
Asterisk show 43% of cpu usuage when there is only one call going. It
actually flactuates down to 11% and up to 43%.

Please guide me as to where to look at?


PID USER  PR  NI  VIRT  RES  SHR S %CPU %MEMTIME+  COMMAND
 2111 asterisk  17   0 40992  17m 8064 S 43.5  1.0 853:53.66 asterisk


Thanks,
Bruce
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Re: [asterisk-users] RTP over TCP

[bruce bruce]

Adobe Air and Adobe FMS are good examples of VoIP working flawlessly over
TCP. We are actually developing a flash phone which needs only TCP to
transmit both signal and audio.

-Bruce

On Sat, Apr 24, 2010 at 2:01 PM, Zeeshan Zakaria  wrote:

> RTP stands for Real-Time Transport Protocol. TCP is not designed to deal
> with real-time data transfer as it takes time to acknowledge packets and
> re-send them if missing. All audio video data transfer happens in real time,
> and it doesn't make any sense to retransmit missing packets. Real time
> packets mixed with old missing packets which are submitted would result in
> an non-understandable audio and video. So how come any system can use TCP
> for real time data transfer, while assuring the quality at the same time.
> Does their exist any such system? I would certainly like to try it, maybe
> they are doing it right using some trick which I don't know yet.
>
> Zeeshan A Zakaria
>
> --
> Sent from my Android phone with K-9 Mail.
>
> On 2010-04-24 1:48 PM, "David Backeberg"  wrote:
>
> On Fri, Apr 23, 2010 at 3:21 PM,  wrote:
> > i have to put an * between two other SIP ga...
>
> Don't do it.
>
> There have been any number of posts to asterisk-users begging asterisk
> to bend over backwards to accommodate Microsoft's foray into the world
> of VoIP. Nobody seems to be asking Microsoft to build a stack
> compatible with the rest of the world of VoIP.
>
> I disagree that sending SIP over TCP is superior to sending it over
> UDP. Think about it for a second. UDP is 'unreliable' in that lost
> packets are not rebroadcast.
>
> Now let's say you have an 'unreliable' connection where it's just
> barely good enough that the SIP call setup goes through, but the RTP
> stream immediately fails.
>
> Why would that be superior to having the SIP call setup getting
> dropped? The end result of no reliable voice is the same, but now you
> have a funkier debug condition that's going to be more complex to
> track down. I personally think it would be superior to see the bad
> connection as early in call setup as possible.
>
> And of course, SIP over UDP is the way the rest of the world works. If
> anybody wants to speak up about a framework that supports BOTH SIP
> over UDP AND SIP over TCP, maybe somebody already built a middleware
> layer that will let Microsoft join the world of voip.
>
>
> --
> _
> -- Bandwidth and Colocati...
>
>
> --
> _
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[asterisk-users] More efficient dial plan for a list of selective inbound numbers

[bruce bruce]

I have a list of CLIDs prefixes that I want to use in a context.

Basically, I want to do this but the list of prefix numbers is much longer.
List of prefixes (556,557,557,989.)

[custom-inbound]
exten => _556,1,answer
exten => _556,n,playback(beep)

exten => _557,1,answer
exten => _557,n,playback(beep)

exten => _558,1,answer
exten => _558,n,playback(beep)

exten => _989,1,answer
exten => _989,n,playback(beep)

If there are like 100s of different prefixes, this list gets really big. Not
desired. How can I have a more efficient dialplan?

Thanks,
Bruce
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Re: [asterisk-users] Portech MV-374 does not register behind NAT

[bruce bruce]

Take out the router/firewall and connect directly to the net to test your
NAT problem theory.

On Thu, Apr 22, 2010 at 12:15 PM, Jonas Kellens wrote:

>  Jared,
>
> thank you for your answer.
>
> As I said in my previous mail, I'm using a Zyxel NBG-419 router (which
> normally supports VoIP and QoS). Firewall is disabled on the Zyxel.
>
> The MV-374 only accepts IP-address, not a FQDN. Will give it another try
> though...
>
> The answer from Portech-support : "use STUN".
>
> Even if the NAT rewrites the IP-address/port combination, why is it a
> problem for the Portech and not for the IP-phones (Grandstream & Snom) ?
> They all communicate on port 5060 --> 5064 (several SIP-accounts)
>
>
> Jonas.
>
>
> Jared Smith wrote:
>
> On Thu, 2010-04-22 at 17:45 +0200, Jonas Kellens wrote:
>
>
>  All goes well when the gateway is connected directly to the
> internet... It's when it is behind NAT the 401 is sent from
> Asterisk...
>
>
>  Is the device registering to an IP address, or do a DNS name?  What type
> of NAT firewall are you using?
>
> This reminds me of a problem I had years ago with a Cisco PIX firewall,
> where it would rewrite IP addresses in the SIP Request URI, causing the
> authentication to fail.  One solution was to have it register to a
> fully-qualified domain name instead of an IP address, so that the
> Request URI wouldn't get overwritten.
>
> It's certainly worth a shot...
>
> --
> Jared Smith
> Digium, Inc.
>
>
>
>
>
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Re: [asterisk-users] Portech MV-374 does not register behind NAT

[bruce bruce]

Try reseting the Gateway (soft reset of the settings) and use only IE to do
the setup again. Nothing else comes to my mind.

Also, create a simple extension in Asterisk or if you are using FreePBX you
don't need to tamper with any ports stuff.


-Bruce

On Thu, Apr 22, 2010 at 3:37 AM, Jonas Kellens wrote:

>  When I comment out the port-parameter (then it defaults to 5060), it is
> still the same...
>
> [Apr 22 09:32:49]
> <--- Transmitting (NAT) to my_pub_ip:5064 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP 192.168.1.23:5064;branch=z9hG4bKc46696a2b5;received=
> my_pub_ip
> From: "SIM 3-1" ;tag=0a99a41b
> To: "SIM 3-1" ;tag=as461d5769
> Call-ID: 5c580e091901a03c39becff71e477...@192.168.1.23
> CSeq: 89 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> WWW-Authenticate: Digest algorithm=MD5, realm="103001vc", nonce="03e68412"
> Content-Length: 0
>
>
> Jonas.
>
> bruce bruce wrote:
>
> Try changing port=5064 to port=5060 in your Asterisk config file. Portech
> will negotiate it's port with Asterisk itself.
>
>
>
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Re: [asterisk-users] Asterisk choking on voice messages announcements

[bruce bruce]

How can I find out what the source of the problem is guys?

As I said I didn't change anything, except for making few minor changes to
the firewall today and that was at Amazon firewall level and not within
CentOS.

What causes these bad dahdi_test values?

P.S. there is only few calls load at anytime on this server.

Thanks

On Wed, Apr 21, 2010 at 8:03 PM, Carlos Chavez wrote:

> On Wed, 2010-04-21 at 19:36 -0400, bruce bruce wrote:
> > Here are result of dahdi_test:
> >
> >
> > [r...@ip-10-251-123-3 ~]# dahdi_test
> > Opened pseudo dahdi interface, measuring accuracy...
> > 99.725% 96.018% 99.532% 91.934% 99.923% 99.923% 99.628% 99.434%
> > -434.763% 99.239% 93.770% 99.141% 99.822% 91.232% 99.727% 93.770%
> > 99.726% -403.227% 98.069% 98.458% 95.136% 98.749% 91.229% 87.622%
> > 98.554% 93.282% -407.620% 94.650% 96.308% 98.750% 96.993% 93.478%
> > 94.063% 93.381% 61.745% -379.400% 99.628% 99.921% 99.142% 96.797%
> > 98.457% 99.337% 87.909% 95.141% -396.880% 99.531% 99.923% 99.921%
> > 91.035% 96.408% 91.916% 90.255% -402.153% 81.079% 74.534% 96.212%
> >
> >
> > What can one tell from these?
> >
> Only that your timing source sucks.  You need 99.9% or higher if
> you
> want a stable system.  I have servers with dahdi_dummy that never go
> below 99.7% accuracy.  You really need to check your timing source.
>
> --
> 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] Asterisk choking on voice messages announcements

[bruce bruce]

I know that anything lower than 99% is bad. But *-400 *?

Anything care of comment?

Thanks,

On Wed, Apr 21, 2010 at 7:45 PM, Steve Howes wrote:

> On 22 Apr 2010, at 00:36, bruce bruce wrote:
> > Opened pseudo dahdi interface, measuring accuracy...
> > 99.725% 96.018% 99.532% 91.934% 99.923% 99.923% 99.628% 99.434%
> > -434.763% 99.239% 93.770% 99.141% 99.822% 91.232% 99.727% 93.770%
> > 99.726% -403.227% 98.069% 98.458% 95.136% 98.749% 91.229% 87.622%
> > 98.554% 93.282% -407.620% 94.650% 96.308% 98.750% 96.993% 93.478%
> > 94.063% 93.381% 61.745% -379.400% 99.628% 99.921% 99.142% 96.797%
> > 98.457% 99.337% 87.909% 95.141% -396.880% 99.531% 99.923% 99.921%
> > 91.035% 96.408% 91.916% 90.255% -402.153% 81.079% 74.534% 96.212%
> >
> > What can one tell from these?
>
> Thats.. Interesting...
>
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Re: [asterisk-users] Asterisk choking on voice messages announcements

[bruce bruce]

It's running on an Amazon instance. No changes to system made and it was
working find previously.

Here is an output of "top":

[r...@ip-10-251-123-3 ~]# top
top - 19:59:48 up  6:52,  1 user,  load average: 0.78, 0.95, 0.99
Tasks:  49 total,   2 running,  47 sleeping,   0 stopped,   0 zombie
Cpu(s):  0.0%us,  0.0%sy,  0.0%ni, 98.7%id,  0.0%wa,  0.0%hi,  0.0%si,
 1.3%st
Mem:   1740948k total,   399504k used,  1341444k free,   105300k buffers
Swap:   917496k total,0k used,   917496k free,   161544k cached

  PID USER  PR  NI  VIRT  RES  SHR S %CPU %MEMTIME+  COMMAND
1 root  15   0  2132  752  648 S  0.0  0.0   0:00.05 init
2 root  RT   0 000 S  0.0  0.0   0:00.00 migration/0
3 root  34  19 000 S  0.0  0.0   0:00.00 ksoftirqd/0
4 root  RT   0 000 S  0.0  0.0   0:00.00 watchdog/0
5 root  10  -5 000 S  0.0  0.0   0:00.00 events/0
6 root  10  -5 000 S  0.0  0.0   0:00.00 khelper
7 root  11  -5 000 S  0.0  0.0   0:00.00 kthread
9 root  20  -5 000 S  0.0  0.0   0:00.00 xenwatch
   10 root  10  -5 000 S  0.0  0.0   0:00.00 xenbus
   17 root  20  -5 000 S  0.0  0.0   0:00.00 kblockd/0
   19 root  20  -5 000 S  0.0  0.0   0:00.00 kseriod
   52 root  25   0 000 S  0.0  0.0   0:00.00 pdflush
   53 root  15   0 000 S  0.0  0.0   0:00.02 pdflush
   54 root  20  -5 000 S  0.0  0.0   0:00.00 kswapd0
   55 root  20  -5 000 S  0.0  0.0   0:00.00 aio/0
  671 root  10  -5 000 S  0.0  0.0   0:00.19 kjournald
  695 root  10  -5 000 S  0.0  0.0   0:00.00 kauditd
  720 root  18  -4  2380  672  424 S  0.0  0.0   0:00.23 udevd
 1439 root  12  -5 000 S  0.0  0.0   0:00.00 kmpathd/0
 1445 root  12  -5 000 S  0.0  0.0   0:00.00 kmirrord
 1463 root  10  -5 000 S  0.0  0.0   0:00.00 kjournald
 1719 root  17   0  2392  572  288 S  0.0  0.0   0:00.00 dhclient
 1804 root  18   0 10576 1040  752 S  0.0  0.1   0:00.34 rsyslogd
 1808 root  25   0  1772  416  352 S  0.0  0.0   0:00.00 rklogd
 1829 root  15   0  6948 1072  688 S  0.0  0.1   0:00.24 sshd
 1858 root  25   0  2640 1208 1040 S  0.0  0.1   0:00.00 mysqld_safe
 1916 mysql 15   0  118m  19m 4904 S  0.0  1.1   0:00.47 mysqld
 1957 root  15   0  9480 1860  784 S  0.0  0.1   0:00.00 sendmail
 1967 smmsp 18   0  8260 1488  632 S  0.0  0.1   0:00.00 sendmail
 1976 root  18   0 24728 7612 4636 S  0.0  0.4   0:00.11 httpd
 1992 root  18   0  3072 1128  584 S  0.0  0.1   0:00.00 crond
 2005 asterisk  18   0 25476 7296 3568 S  0.0  0.4   0:00.09 httpd
 2006 asterisk  15   0 25496 7300 3556 S  0.0  0.4   0:00.04 httpd
 2007 asterisk  15   0 25816 7364 3596 S  0.0  0.4   0:00.11 httpd
 2008 asterisk  20   0 29348 9876 4432 S  0.0  0.6   0:00.04 httpd
 2009 asterisk  15   0 24888 5244 2092 S  0.0  0.3   0:00.09 httpd
 2010 asterisk  17   0 25496 7300 3540 S  0.0  0.4   0:00.08 httpd
 2011 asterisk  17   0 25480 7344 3572 S  0.0  0.4   0:00.07 httpd
 2012 asterisk  15   0 25496 7252 3516 S  0.0  0.4   0:00.03 httpd


On Wed, Apr 21, 2010 at 7:56 PM, Sean Brady  wrote:

>
>
> On 04/21/2010 05:36 PM, bruce bruce wrote:
>
> Here are result of dahdi_test:
>
>  [r...@ip-10-251-123-3 ~]# dahdi_test
> Opened pseudo dahdi interface, measuring accuracy...
> 99.725% 96.018% 99.532% 91.934% 99.923% 99.923% 99.628% 99.434%
> -434.763% 99.239% 93.770% 99.141% 99.822% 91.232% 99.727% 93.770%
> 99.726% -403.227% 98.069% 98.458% 95.136% 98.749% 91.229% 87.622%
> 98.554% 93.282% -407.620% 94.650% 96.308% 98.750% 96.993% 93.478%
> 94.063% 93.381% 61.745% -379.400% 99.628% 99.921% 99.142% 96.797%
> 98.457% 99.337% 87.909% 95.141% -396.880% 99.531% 99.923% 99.921%
> 91.035% 96.408% 91.916% 90.255% -402.153% 81.079% 74.534% 96.212%
>
>  What can one tell from these?
>
> On Wed, Apr 21, 2010 at 6:59 PM, bruce bruce  wrote:
>
>> Thanks for the input.
>>
>> I am going to check this once I get access to system again tonight.
>>
>> But I thought the timing source dahdi_dummy is only good for features like
>> MeetMe or conference rooms? or am I wrong and it has an effect on any type
>> of calls and checking voice messages?
>>
>> Thanks
>>
>>   On Wed, Apr 21, 2010 at 2:49 PM, Ryan Bullock wrote:
>>
>>>  So I be it sounds like all the recordings are underwater.
>>>
>>> Are you using dahdi for timing? Can you run dahdi_test?
>>>
>>>  Asterisk needs a good timing source, in the case when you don't have a
>>> physical card providing it, it relies on kernel ticks or the RTC (or HPET).
>>> Because of the nature of virtual m

Re: [asterisk-users] Asterisk choking on voice messages announcements

[bruce bruce]

Here are result of dahdi_test:

[r...@ip-10-251-123-3 ~]# dahdi_test
Opened pseudo dahdi interface, measuring accuracy...
99.725% 96.018% 99.532% 91.934% 99.923% 99.923% 99.628% 99.434%
-434.763% 99.239% 93.770% 99.141% 99.822% 91.232% 99.727% 93.770%
99.726% -403.227% 98.069% 98.458% 95.136% 98.749% 91.229% 87.622%
98.554% 93.282% -407.620% 94.650% 96.308% 98.750% 96.993% 93.478%
94.063% 93.381% 61.745% -379.400% 99.628% 99.921% 99.142% 96.797%
98.457% 99.337% 87.909% 95.141% -396.880% 99.531% 99.923% 99.921%
91.035% 96.408% 91.916% 90.255% -402.153% 81.079% 74.534% 96.212%

What can one tell from these?

On Wed, Apr 21, 2010 at 6:59 PM, bruce bruce  wrote:

> Thanks for the input.
>
> I am going to check this once I get access to system again tonight.
>
> But I thought the timing source dahdi_dummy is only good for features like
> MeetMe or conference rooms? or am I wrong and it has an effect on any type
> of calls and checking voice messages?
>
> Thanks
>
> On Wed, Apr 21, 2010 at 2:49 PM, Ryan Bullock  wrote:
>
>> So I be it sounds like all the recordings are underwater.
>>
>> Are you using dahdi for timing? Can you run dahdi_test?
>>
>> Asterisk needs a good timing source, in the case when you don't have a
>> physical card providing it, it relies on kernel ticks or the RTC (or HPET).
>> Because of the nature of virtual machines they don't always get access to
>> the processor when they want and therefore their timing can get skewed and
>> can be bad for real-time applications.
>>
>> There are some patches/work-arounds that you can do. You might want to
>> google 'asterisk in a virtual machine' or 'asterisk timing virutal machine',
>> or anything along those lines.
>>
>> I think I remember in some of the recent dahdi or asterisk release notes
>> that they changed some settings to be more virtual machine friendly. So
>> maybe make sure you are running the latest versions?
>>
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Re: [asterisk-users] Asterisk choking on voice messages announcements

[bruce bruce]

Thanks for the input.

I am going to check this once I get access to system again tonight.

But I thought the timing source dahdi_dummy is only good for features like
MeetMe or conference rooms? or am I wrong and it has an effect on any type
of calls and checking voice messages?

Thanks

On Wed, Apr 21, 2010 at 2:49 PM, Ryan Bullock  wrote:

> So I be it sounds like all the recordings are underwater.
>
> Are you using dahdi for timing? Can you run dahdi_test?
>
> Asterisk needs a good timing source, in the case when you don't have a
> physical card providing it, it relies on kernel ticks or the RTC (or HPET).
> Because of the nature of virtual machines they don't always get access to
> the processor when they want and therefore their timing can get skewed and
> can be bad for real-time applications.
>
> There are some patches/work-arounds that you can do. You might want to
> google 'asterisk in a virtual machine' or 'asterisk timing virutal machine',
> or anything along those lines.
>
> I think I remember in some of the recent dahdi or asterisk release notes
> that they changed some settings to be more virtual machine friendly. So
> maybe make sure you are running the latest versions?
>
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Re: [asterisk-users] Asterisk choking on voice messages announcements

[bruce bruce]

yes, it's on Amazon.

On Wed, Apr 21, 2010 at 2:26 PM, Ryan Bullock  wrote:

> Are you running asterisk in a virtual machine?
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Re: [asterisk-users] Asterisk choking on voice messages announcements

[bruce bruce]

Yes, it's all g.711 ulaw.

On Wed, Apr 21, 2010 at 1:37 PM, Darrick Hartman (lists) <
dhart...@djhsolutions.com> wrote:

> Are your sound files being transcoded or played back in their native
> formats?
>
> On 04/21/2010 12:25 PM, bruce bruce wrote:
> > Hi Everyone,
> >
> > I have a weired situation where calls in and out are proceessed all
> > right but when I dial *97 Asterisk is literally choking when it comes to
> > announcements like "Password" or "Call from 205-456-". Each one of
> > those announcements can take like 10+ seconds to finish with most of it
> > not even compoundable.
> >
> > I run "top" and there is no heavy load on CPU or RAM. I dial out and
> > it's all fine.
> >
> > Can you please give me some pointers as to where to look for the problem?
> >
> > Also, if I allow a call to go to voice-mail on my extension, the
> > announcement, "The person at extension 4000 is not available" is also
> > garbled and very slow like a choking sound. This is serious because
> > people think they are have reached a faulty answering machine or just
> > cut off because there is a long instance of silence sometime.
> >
> > Thanks
> >
> >
> --
> Darrick Hartman
> DJH Solutions, LLC
> http://www.djhsolutions.com
>
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[asterisk-users] Asterisk choking on voice messages announcements

[bruce bruce]

Hi Everyone,

I have a weired situation where calls in and out are proceessed all right
but when I dial *97 Asterisk is literally choking when it comes to
announcements like "Password" or "Call from 205-456-". Each one of those
announcements can take like 10+ seconds to finish with most of it not even
compoundable.

I run "top" and there is no heavy load on CPU or RAM. I dial out and it's
all fine.

Can you please give me some pointers as to where to look for the problem?

Also, if I allow a call to go to voice-mail on my extension, the
announcement, "The person at extension 4000 is not available" is also
garbled and very slow like a choking sound. This is serious because people
think they are have reached a faulty answering machine or just cut off
because there is a long instance of silence sometime.

Thanks
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Re: [asterisk-users] Portech MV-374 does not register

[bruce bruce]

Try changing port=5064 to port=5060 in your Asterisk config file. Portech
will negotiate it's port with Asterisk itself.

On Tue, Apr 20, 2010 at 10:50 AM, Jonas Kellens wrote:

>  When there is a register with bad password, then this is the SIP response
> :
>
> <>
> [Apr 20 16:44:29]
> <--- Transmitting (NAT) to my_public_ip:5061 --->
> *SIP/2.0 403 Forbidden (Bad auth)*
> Via: SIP/2.0/UDP 192.168.1.22:5061
> ;branch=z9hG4bKbdc15a66904d1239;received=my_public_ip
> From: "test" ;tag=e1b2b6acd8aca5f0
> To: ;tag=as1d7ac6a5
> Call-ID: b5f62f9b0ec59...@192.168.1.22
> CSeq: 20002 REGISTER
>
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> <>
> [Apr 20 16:44:29] NOTICE[2541]: chan_sip.c:15889 handle_request_register:
> Registration from '' failed for
> 'my_public_ip' - Wrong password
>
>
> This is quite different from the "wrong registrations" I get from the
> MV-374 gateway...
>
> Don't know why local registration succeeds and to a public server fails...
> NAT anyone ? But then why does it work like a charm with an IP-phone
> (Grandstream) ?!
>
>
> Jonas.
>
>
>
> bruce bruce wrote:
>
> I have had problems with Portech firmware using Chrome browser. The problem
> was that when I changed the password on the gateway it would apply that
> password to SIP PEERS as well. So, maybe, you are actually not having the
> right password in your SIP peer as well and hence your Asterisk sends
> Unauthorized.
>
>
>
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Re: [asterisk-users] Portech MV-374 does not register

[bruce bruce]

I have had problems with Portech firmware using Chrome browser. The problem
was that when I changed the password on the gateway it would apply that
password to SIP PEERS as well. So, maybe, you are actually not having the
right password in your SIP peer as well and hence your Asterisk sends
Unauthorized.

-Bruce

On Tue, Apr 20, 2010 at 9:29 AM, Jonas Kellens wrote:

>  With tcpdump I saw that there were packets coming in from the GSM-gateway
> to the public Asterisk-server.
> I saw nothing on the Asterisk-CLI that told me that there were attempts to
> register, but a "sip debug" shows this :
>
> <>
> [Apr 20 15:07:41] Scheduling destruction of SIP dialog '
> 0cd637c143e6667c4b5279b713b50...@192.168.1.25' in 32000 ms (Method:
> REGISTER)
> [Apr 20 15:07:41] Really destroying SIP dialog '
> 5c4fc9a47a3f5d545608747f45186...@192.168.1.25' Method: REGISTER
> [Apr 20 15:07:41]
> <--- SIP read from my_public_ip:5066 --->
> REGISTER sip:my_asterisk_ip SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.25:5066;rport;branch=z9hG4bK959181a47a
> From: "SIM 3" ;tag=4c9ddc99
> To: "SIM 3" 
> Call-ID: 001816f82d8bf3d43f34faa82f344...@192.168.1.25
> Contact: 
> CSeq: 2354 REGISTER
> Max-Forwards: 70
> Expires: 60
> Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE
> User-Agent: Mv-37x (904290)
> Content-Length: 0
>
>
> <->
> [Apr 20 15:07:41] --- (12 headers 0 lines) ---
> [Apr 20 15:07:41] Using latest REGISTER request as basis request
> [Apr 20 15:07:41] Sending to my_public_ip : 5066 (NAT)
> [Apr 20 15:07:41]
> <--- Transmitting (NAT) to my_public_ip:5066 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 192.168.1.25:5066
> ;branch=z9hG4bK959181a47a;received=my_public_ip;rport=5066
> From: "SIM 3" ;tag=4c9ddc99
> To: "SIM 3" 
> Call-ID: 001816f82d8bf3d43f34faa82f344...@192.168.1.25
> CSeq: 2354 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> <>
> [Apr 20 15:07:41]
> <--- Transmitting (NAT) to my_public_ip:5066 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP 192.168.1.25:5066
> ;branch=z9hG4bK959181a47a;received=my_public_ip;rport=5066
> From: "SIM 3" ;tag=4c9ddc99
> To: "SIM 3" ;tag=as09b99e8c
> Call-ID: 001816f82d8bf3d43f34faa82f344...@192.168.1.25
> CSeq: 2354 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> WWW-Authenticate: Digest algorithm=MD5, realm="103001vc", nonce="3c911c4a"
> Content-Length: 0
>
> How come there is a register attempt that is "Unauthorized" and how come
> this doesn't show on the CLI ??
>
>
> Kind regards,
>
> Jonas.
>
>
> Jonas Kellens wrote:
>
> Hello list,
>
> has anyone experience with the Portech MV-374 GSM-gateway ?
>
> I'm trying to register the SIP-accounts to a public SIP-server but that
> fails.
>
> When trying to register to a local Asterisk-server, all goes well.
>
> So anyone knows what special setting I need to make to register my
> SIP-accounts/SIM-cards to a public IP ??
>
>
>
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Re: [asterisk-users] Zap PRI failed with Cause 34 - Where to check for problems?

[bruce bruce]

I have already done an "amportal restart" which restarted Asterisk and
everything works fine now.

Here is the output of "pri show span 1":

Primary D-channel: 24
Status: Provisioned, Up, Active
Switchtype: National ISDN
Type: CPE
Window Length: 0/7
Sentrej: 0
SolicitFbit: 1
Retrans: 0
Busy: 0
Overlap Dial: 0
Logical Channel Mapping: 0
T200 Timer: 1000
T203 Timer: 1
T305 Timer: 3
T308 Timer: 4000
T309 Timer: -1
T313 Timer: 4000
N200 Counter: 3


thanks,
Bruce

On Mon, Apr 19, 2010 at 3:00 PM, Doug Lytle  wrote:

> bruce bruce wrote:
> >
> > [2010-04-19 08:45:50] WARNING[29707] app_dial.c: Unable to create
> > channel of type 'ZAP' (cause 34 - Circuit/channel congestion)
> >
> >
> > Where can I look to debug the problem?
> >
>
> What does the:
>
> pri show span 1
>
> Output look like?
>
> Doug
>
> --
>
> Ben Franklin quote:
>
> "Those who would give up Essential Liberty to purchase a little Temporary
> Safety, deserve neither Liberty nor Safety."
>
>
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Re: [asterisk-users] X-lite direct sip call - Is it possible?

[bruce bruce]

But you are definitely right with the sjphone being very straight forward
with direct peer-peer dials. I had to put many minutes into setting x-lite
to do direct peer-peer calls.

G'day,
Bruce

On Mon, Apr 19, 2010 at 11:29 AM, Alyed  wrote:

> Sorry for my mislead should have said "I've never been able to with xlite"
> it's just with Sjphone it's straight forward.
>
> Alyed
>
> 2010/4/19 bruce bruce 
>
> That is not correct. It's possible by adding a display name and adding the
>> IP address of the pbx you are calling as the host ip. Then uncheck the
>> register button and place calls in format: ex...@ip
>>
>> For adding characters and @ sign, push "space bar" and then type whatever
>> you wish.
>>
>> -Bruce
>>
>> On Mon, Apr 19, 2010 at 12:08 AM, Alyed  wrote:
>>
>>> You can't do that with Xlite, try Sjphone instead.
>>>
>>> Alyed
>>>
>>>
>>> 2010/4/17 bruce bruce 
>>>
>>>> Hi Guys,
>>>>
>>>> Wondering if anyone has tried to make a direct SIP peer to peer call
>>>> using x-lite without any registrations of any sort. I can't seem to find 
>>>> the
>>>> setting.
>>>>
>>>> Thanks,
>>>> bruce
>>>>
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>>>>
>>>
>>>
>>>
>>>
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>>
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>
>
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[asterisk-users] Zap PRI failed with Cause 34 - Where to check for problems?

[bruce bruce]

Hello Everyone,

I have a system that was working on Sunday 1 P.M. and then gives Congestion
on Monday morning. Sometimes over night it probably stopped working. It's a
PBXinaFLASH with Asterisk 1.4 and libPRI with a 23 channel PRI connected and
24th D-Channel.

This is all I see in /var/log/asterisk/full:

[2010-04-19 08:45:50] WARNING[29707] app_dial.c: Unable to create channel of
type 'ZAP' (cause 34 - Circuit/channel congestion)


Where can I look to debug the problem?

Thanks,
Bruce
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Re: [asterisk-users] X-lite direct sip call - Is it possible?

[bruce bruce]

That is not correct. It's possible by adding a display name and adding the
IP address of the pbx you are calling as the host ip. Then uncheck the
register button and place calls in format: ex...@ip

For adding characters and @ sign, push "space bar" and then type whatever
you wish.

-Bruce

On Mon, Apr 19, 2010 at 12:08 AM, Alyed  wrote:

> You can't do that with Xlite, try Sjphone instead.
>
> Alyed
>
>
> 2010/4/17 bruce bruce 
>
>> Hi Guys,
>>
>> Wondering if anyone has tried to make a direct SIP peer to peer call using
>> x-lite without any registrations of any sort. I can't seem to find the
>> setting.
>>
>> Thanks,
>> bruce
>>
>> --
>> _
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>
>
>
>
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