Re: [asterisk-users] SSH connection hangs on logout?

[Steve Edwards]

On Thu, 24 Aug 2006, Jeremy McNamara wrote:


Rushowr wrote:

Hey all, I have an interesting issue that just recently started when I
grabbed a copy of the trunk about a week ago and compiled it. Ever since
that compile, if I start Asterisk (disconnected terminal, using
safe_asterisk to launch) and then continue on about my work with it, when I
disconnect my SSH terminal (using latest version of PuTTY) the session no
longer closes it just hangs. I've even changed the Putty setting to close
the window even on unclean exit but it still hangs the connection... I had
something similar once with Zabbix a while back, but never Asterisk.

Anyone else experience this?


Start asterisk using  safe_asterisk or via asterisk -f

I prefer the safe_asterisk shell script, since if asterisk seg faults, there 
is a good chance asterisk will get automatically restarted.


Jeremy McNamara
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You may need to redirect stdin, stdout, stderr like:

run_asterisk\
0/dev/null\
1/dev/null\
2/dev/null\


Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] AstLinux 0.4.3 Released!

[Kristian Kielhofner]

Hello everyone,

I have released AstLinux 0.4.3:

http://sourceforge.net/projects/astlinux/

	For all of those that have been waiting to switch to 0.4.x, this is 
your chance.  The few remaining problems with uclibc have been fixed 
(i.e. voicemail timezones and voicemail - email via MSMTP).


	Don't forget to peek around in SVN for all kinds of goodies. 
Especially trunk - the Gumstix is now a direct target for builds. 
That's right, build AstLinux for a Gumstix just as easily as a Soekris!


--
Kristian Kielhofner
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Re: [asterisk-users] SSH connection hangs on logout?

[Tzafrir Cohen]

On Wed, Aug 23, 2006 at 11:03:23PM -0700, Steve Edwards wrote:
 On Thu, 24 Aug 2006, Jeremy McNamara wrote:
 
 Rushowr wrote:
 Hey all, I have an interesting issue that just recently started when I
 grabbed a copy of the trunk about a week ago and compiled it. Ever since
 that compile, if I start Asterisk (disconnected terminal, using
 safe_asterisk to launch) and then continue on about my work with it, when 
 I
 disconnect my SSH terminal (using latest version of PuTTY) the session no
 longer closes it just hangs. I've even changed the Putty setting to close
 the window even on unclean exit but it still hangs the connection... I had
 something similar once with Zabbix a while back, but never Asterisk.
 
 Anyone else experience this?
 
 Start asterisk using  safe_asterisk or via asterisk -f
 
 I prefer the safe_asterisk shell script, since if asterisk seg faults, 
 there is a good chance asterisk will get automatically restarted.
 
 Jeremy McNamara
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 You may need to redirect stdin, stdout, stderr like:
 
 run_asterisk\
 0/dev/null\
 1/dev/null\
 2/dev/null\
 
 

In other words: 

A plain 'asterisk' (without '-c' and such) that daemonizes and does
exactly that for you, among others.

Asterisk is a daemon, rather than an interactive program. Thus its
handling for SIGHUP is to re-read configuration rather than detach from
the terminal.

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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[asterisk-users] monitor a hangup in AGI application?

[Abubakar A. Khaliq]

Hello,

If a call is connected to an AGI application, the application terminates if
the call is hang-up.

I have an AGI application in which I want to do a certain operation while
call hangs up. Is it possible for my application to know when a call is
hang-up? Is there any way to monitor this scenario?

Also, is it possible to know the channel created for agi application?
I mean, when a cal is connected to an AGI application, it has two channels.
One channel is created for call and the other is created for AGI
application. Is it possible for me to know the channel created for my AGI
application?

Eagerly waiting for reply.

Regards,
Abubakar A. Khaliq




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Re: [asterisk-users] Annoying Bristuff

[Kai Ober]

Does anyone have any other tips.


use mISDN ;)

or are you bound to bristuff because you need speciall features of this?

KAi
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RE: [asterisk-users] Unable to match on CallerID in an include block

[Steve Hanselman]

I'll run some more tests but it's not very different from the posting?

Steve


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian
Lyndon-Smith
Sent: 22 August 2006 18:22
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Unable to match on CallerID in an include
block

I suspect that your dialplan is more than you show ;)

It works just fine for me with svn trunk

[from-sip]

include = common

[common]
exten = 1234,1,NoOp(Hmm ${CALLERID(num)})

exten = 1234/7708,1,NoOp(Here)

If I dial 1234 from my 7708 extension, I get the NoOp(Here)
If I dial 1234 from my 7701 extension, I get the NoOp(Hmm 7701)

Julian.

Steve Hanselman wrote:
 Hi Julian,

 Ah, a very good point, I put that in my first cut but had completely
 forgotten in this one!

 1.2.10

 Steve


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Julian
 Lyndon-Smith
 Sent: 22 August 2006 17:30
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Unable to match on CallerID in an
include
 block

 What version of asterisk ?

 Julian

 Steve Hanselman wrote:
 Is there any reason why I can't use the xxx/callerid format in an
 include section?



 It doesn't seem to work, but if I paste the lines into the main
 section
 where I include the block it does?







 E.g. this doesn't work



 [telewest]



 Include = spamblock



 [spamblock]



 _X./12345,s,macro(spamcall)



 Whereas this does:



 [telewest]



 _X./12345,s,macro(spamcall)





 Any ideas?



 Steve



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[asterisk-users] channel variable

[unplug]

Hi,

 I have set some variable in a call.
Set(testmode=1)
 For some reason, such as forward the call, the follow command called.
Dial(Local/1234567)
 It will go through the dial plan again but the value of variable
testmode is nothing instead of 1.

How can I maintain the value of the variable in the above case?

unplug
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Re: [asterisk-users] Annoying Bristuff

[Andrew Nowrot]

or are you bound to bristuff because you need speciall features of this?Well you are right. Bristuff has more features than mISDN.After loading the florz patch messages in kernlog turn into this
Aug 24 10:18:08 asterisk kernel: zaphfc[0]: received d channel frame with bad CRC.Aug 24 10:20:45 asterisk kernel: zaphfc[0]: received d channel frame with bad CRC.Aug 24 10:21:40 asterisk last message repeated 2 times
Aug 24 10:25:18 asterisk kernel: zaphfc[0]: received d channel frame with bad CRC.Aug 24 10:26:19 asterisk last message repeated 2 timesAug 24 10:26:49 asterisk kernel: zaphfc[0]: received d channel frame with bad CRC.
Aug 24 10:28:20 asterisk last message repeated 3 timesAug 24 10:28:50 asterisk kernel: zaphfc[0]: received d channel frame with bad CRC.Aug 24 10:29:51 asterisk last message repeated 2 timesAug 24 10:30:51 asterisk kernel: zaphfc[0]: empty HDLC frame received.
Aug 24 10:30:52 asterisk kernel: zaphfc[0]: received d channel frame with bad CRC.Aug 24 10:31:22 asterisk kernel: zaphfc[0]: received d channel frame with bad CRC.Aug 24 10:32:22 asterisk last message repeated 2 times
Aug 24 10:32:53 asterisk kernel: zaphfc[0]: received d channel frame with bad CRC.Aug 24 10:33:53 asterisk last message repeated 2 timesAug 24 10:35:18 asterisk last message repeated 2 timesAug 24 10:36:19 asterisk last message repeated 2 times
Aug 24 10:37:19 asterisk kernel: zaphfc[0]: empty HDLC frame received.Aug 24 10:37:19 asterisk kernel: zaphfc[0]: received d channel frame with bad CRC.Aug 24 10:38:20 asterisk kernel: zaphfc[0]: received d channel frame with bad CRC.
CheersAndrew
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[asterisk-users] Re: channel variable

[Tony Mountifield]

In article [EMAIL PROTECTED],
unplug [EMAIL PROTECTED] wrote:
 Hi,
 
   I have set some variable in a call.
 Set(testmode=1)
   For some reason, such as forward the call, the follow command called.
 Dial(Local/1234567)
   It will go through the dial plan again but the value of variable
 testmode is nothing instead of 1.
 
 How can I maintain the value of the variable in the above case?

Try using Set(__testmode=1), but still refer to it as ${testmode}.

The __ tells Asterisk to propagate the variable to created channels.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [asterisk-users] About IVR and Oracle

[Tim Panton]

On 23 Aug 2006, at 23:07, Javier Lara Sanchez wrote:


Dear All,



I need to buid an IVR that could make a request to a data base  
(oracle) in a remote host.




 The idea is that an user dial a extension with 2 options and one  
of them ask for a data (in the case a date). This data is the field  
that the data base needs to find the information that the user are  
looking for..




Somebody know if this is posible or have any idea where can I find  
information about this?


We have done this by using the (excellent) asterisk-java api on  
sourceforge to create a

FAstAGI server on the database box. (http://asterisk-java.org/latest/)

Asterisk makes a FastAGI call to this server, the server looks up the  
query by JDBC, sets
the result in a channel variable and returns control to asterisk's  
dialplan.

The dialplan then does the 'right' thing with the call.

This means we keep the connection duration to the FAstAgi server to a  
minimum.


Tim.



Thank

Regard

Javier









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Tim Panton

www.mexuar.com



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[asterisk-users] Re: Nokia E60/61/70 and SIP

[Benny Amorsen]

 H == Haspers  [EMAIL PROTECTED] writes:

H We are using some E61 and E70's with asterisk. Only problem we have
H at this moment is that we are unable to use a password for the
H authentication. I haven't found out yet why this isn't working.
H They are working good, but I would like to see some small things
H changed in future firmware versions (like being able to select
H multiple WLAN points (Access groups) instead of just one.

E70 works with passwords here. No trouble.

The main issue is that the E70 can't automatically switch to cellular
when the phone is out of WLAN coverage. It is a bit silly to have to
click option-s

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Re: [asterisk-users] Re: Nokia E60/61/70 and SIP

[Simon Woodhead]

Hi Benny,The E61 handles this just fine. With SIP as the default channel to dial and no WiFi coverage, you get a message asking if you'd like to dial by cellular. Works nicely other than a few stability issues.
SimonOn 24 Aug 2006 11:23:39 +0200, Benny Amorsen [EMAIL PROTECTED] wrote:
 H == Haspers[EMAIL PROTECTED] writes:H We are using some E61 and E70's with asterisk. Only problem we haveH at this moment is that we are unable to use a password for the
H authentication. I haven't found out yet why this isn't working.H They are working good, but I would like to see some small thingsH changed in future firmware versions (like being able to select
H multiple WLAN points (Access groups) instead of just one.E70 works with passwords here. No trouble.The main issue is that the E70 can't automatically switch to cellularwhen the phone is out of WLAN coverage. It is a bit silly to have to
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[asterisk-users] Re: Nokia E60/61/70 and SIP

[Benny Amorsen]

Sorry for the duplicate post.

 H == Haspers  [EMAIL PROTECTED] writes:

H We are using some E61 and E70's with asterisk. Only problem we have
H at this moment is that we are unable to use a password for the
H authentication. I haven't found out yet why this isn't working.
H They are working good, but I would like to see some small things
H changed in future firmware versions (like being able to select
H multiple WLAN points (Access groups) instead of just one.

E70 works with passwords here. No trouble.

The main issue is that the E70 can't automatically switch to cellular
when the phone is out of WLAN coverage. It is a bit silly to have to
click options-call-voice call.


/Benny


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Re: [asterisk-users] Re: channel variable

[unplug]

Do you mean it is a global variable instead of channel variable?



Try using Set(__testmode=1), but still refer to it as ${testmode}.

The __ tells Asterisk to propagate the variable to created channels.

Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[asterisk-users] Monitoring/Listening In

[Scott Pinhorne]

Hi

I wish to setup asterisk for training purposes so that I am able to 
listen in to an extension while a call is going on?


Has anyone done this?

Thanks
SP
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RE: [asterisk-users] Re: Nokia E60/61/70 and SIP

[Haspers]

Strange,

What settings do you use? I followed this link
http://www.newlc.com/Using-SIP-with-Nokia-Series60-and.html but without any
luck. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Benny Amorsen
Sent: donderdag 24 augustus 2006 11:24
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Nokia E60/61/70 and SIP

 H == Haspers  [EMAIL PROTECTED] writes:

H We are using some E61 and E70's with asterisk. Only problem we have 
H at this moment is that we are unable to use a password for the 
H authentication. I haven't found out yet why this isn't working.
H They are working good, but I would like to see some small things 
H changed in future firmware versions (like being able to select 
H multiple WLAN points (Access groups) instead of just one.

E70 works with passwords here. No trouble.

The main issue is that the E70 can't automatically switch to cellular when
the phone is out of WLAN coverage. It is a bit silly to have to click
option-s

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Re: [asterisk-users] Re: Nokia E60/61/70 and SIP

[Thomas Kenyon]

Simon Woodhead wrote:
 Hi Benny,

 The E61 handles this just fine. With SIP as the default channel to
 dial and no WiFi coverage, you get a message asking if you'd like to
 dial by cellular. Works nicely other than a few stability issues.

The E60 appears to handle WPA2 fine, with roaming across the access
points not being a problem.

The only problem I have been having, is that during the day it will just
disconnect from the wireless network and require power cycling.

 Simon

 On 24 Aug 2006 11:23:39 +0200, *Benny Amorsen*
 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:

  H == Haspers  [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
 writes:

 H We are using some E61 and E70's with asterisk. Only problem we have
 H at this moment is that we are unable to use a password for the
 H authentication. I haven't found out yet why this isn't working.
 H They are working good, but I would like to see some small things
 H changed in future firmware versions (like being able to select
 H multiple WLAN points (Access groups) instead of just one.

 E70 works with passwords here. No trouble.

 The main issue is that the E70 can't automatically switch to cellular
 when the phone is out of WLAN coverage. It is a bit silly to have to
 click option-s

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Re: [asterisk-users] Re: Nokia E60/61/70 and SIP

[Simon Woodhead]

Hi Haspers,Makes sure you have created an 'Internet tel' profile. It doesn't appear to do anything but was vital in getting it working for me. The other settings in the how to look sensible.Simon
On 8/24/06, Haspers [EMAIL PROTECTED] wrote:
Strange,What settings do you use? I followed this linkhttp://www.newlc.com/Using-SIP-with-Nokia-Series60-and.html but without anyluck.
-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]
] On Behalf Of Benny AmorsenSent: donderdag 24 augustus 2006 11:24To: asterisk-users@lists.digium.comSubject: [asterisk-users] Re: Nokia E60/61/70 and SIP
 H == Haspers[EMAIL PROTECTED] writes:H We are using some E61 and E70's with asterisk. Only problem we haveH at this moment is that we are unable to use a password for the
H authentication. I haven't found out yet why this isn't working.H They are working good, but I would like to see some small thingsH changed in future firmware versions (like being able to select
H multiple WLAN points (Access groups) instead of just one.E70 works with passwords here. No trouble.The main issue is that the E70 can't automatically switch to cellular whenthe phone is out of WLAN coverage. It is a bit silly to have to click
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Re: [asterisk-users] IAX2 extn not registering on 4569

[Thomas Kenyon]

[EMAIL PROTECTED] wrote:
 Hi all,
  
 Just having a strange situation with no clues how to solve.
  
 I have an Asterisk/TRIXBOX located in US and an IAX extn running on
 PA168V ATA in another country. All my configs seems to be on 4569 but
 i see my extn connected at a different port like 13569.
  
 How can i make it to register at 4569 on my asterisk?
Are you sur eyou're not looking at the source port instead?

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[asterisk-users] Multiple lines in body of UserEvent

[Florian Muellner]

Hi everybody,

I'm trying to send a user event from the dialplan like this:

exten = s,n,UserEvent(EventName|var1:value1^var2:value2)

The event is sent just fine, but the body is not split in two lines as 
it should be according to 
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+UserEvent.
Using 1.2.9.1 here, someone knows whether the split character has 
changed or the feature has been removed and it is simply not possible 
(anymore? Didn't need to try with previous versions...) to split the 
body argument of user events?


Gratefull for any hint,
Florian Müllner
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Re: [asterisk-users] Monitoring/Listening In

[Time Bandit]

I wish to setup asterisk for training purposes so that I am able to
listen in to an extension while a call is going on?


http://www.voip-info.org/wiki-Asterisk+cmd+ZapBarge
and
http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy
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[asterisk-users] quintum Calling Card

[Abdul]

Hi all,Could anyone provide me some usefull link or some idea, how to configure quintum as calling card purpose with Asterisk.Already i created AGI script which working with SIPURA well. But i do not have the idea about quintum how to configure so quintum will dial our asterisk calling card number.i have add [EMAIL PROTECTED] server, so if some one trying to call FXO line of quintum then quintum should dial automatically this URI and rest my AGI will do. even i don't wnat to use quintum IVR.I will be appriciate for your helps.Regards 
		Stay in the know. Pulse on the new Yahoo.com.  Check it out. 
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RE: [asterisk-users] quintum Calling Card

[Jonathan k. Creasy]

Abdul, it doesnt sound like you
need to do anything to the Quintum. I would recommend making your dial plan execute
the AGI script of your choice no matter what number is dialed from the context
where the quantum users land. 



-Jonathan













From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Abdul
Sent: Thursday, August 24, 2006
8:12 AM
To:
asterisk-users@lists.digium.com
Subject: [asterisk-users] quintum
Calling Card





Hi all,

Could anyone provide me some usefull link or some idea, how to configure
quintum as calling card purpose with Asterisk.

Already i created AGI script which working with SIPURA well. But i do not have
the idea about quintum how to configure so quintum will dial our asterisk
calling card number.

i have add [EMAIL PROTECTED] server, so if some one trying to call FXO line of
quintum then quintum should dial automatically this URI and rest my AGI will
do. even i don't wnat to use quintum IVR.

I will be appriciate for your helps.

Regards

 







Stay in the know. Pulse on the new Yahoo.com. Check it
out. 








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[asterisk-users] Active Directory Listing Feauture

[Mohamed A. Gombolaty]

Dear All,
I am currently very stumped on the subject of Active Directory listing,
as I am unable to find any documents regarding this feature thus I am unable
to configure it or know how to use it. Does anyone have any useful info
or documents regarding this feature in terms of how to or guides I will
be very much thankful.


--
Thx
MAG

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Re: [asterisk-users] Active Directory Listing Feauture

[Doug Lytle]

Mohamed A. Gombolaty wrote:

Dear All,

I am currently very stumped on the subject of Active Directory 
listing, as I am unable to find any documents regarding this feature 
thus I am




http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Directory

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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[asterisk-users] E61

[Dovid Bender]

Anyone here use the Nokia E61 ? I am looking to 
invest in a wifi phone and I want to get the best. Is it good as far as 
reception ? That is of most importance to me. Thanks.

Dovid
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RE: [asterisk-users] E61

[Andreas Sikkema]

 Anyone here use the Nokia E61 ? I am looking to invest in a 
 wifi phone and I want to get the best. Is it good as far as 
 reception ? That is of most importance to me. Thanks.

I've tried it in the last couple of days. The biggest issue for 
me ist that it HAS to be on the same side of a NAT as the 
server it talks to (asterisk, ser, etc). If it is on the 
private side of a NAT and the server is on the public side, it 
doesn't work. I've read something on the Nokia forums that 
Nokia is aware of the problem and it will be solved.

My problem is that they want to solve this using STUN etc, 
while I would prefer they also wouldn't have the software 
care if it is on the inside of a NAT like most other CPE's 
so our platform can take care of things.

-- 
Andreas SikkemaBBeyond
Software EngineerPlaneetbaan 4
+31 (0)23 70743422132 HZ Hoofddorp 
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Re: [asterisk-users] SpanDSP Error

[Christian Jensen]

I have not found any solution to the problem I am talking about in  
the archives Steve. I have reinstalled and downgraded from pre26 to  
pre21 of  SpanDSP 0.0.2 and still to no avail. I have followed all  
the instructions and ways of fixing this problem and have found none  
to be a solution.

-configured with prefix /usr
-make
-make install
-then moved the patch file and the corresponding .c files into the  
apps directory of my asterisksource
-Then patched the Makefile and recompiled asterisk. I have done it  
with about 4 versions so far of which I cannot recall which ones.  
Pre26 and Pre21 definately

Anyone?

-Chris
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Re: [asterisk-users] E61

[Simon Woodhead]

Hi Andreas,That is incorrect. It works just fine through NAT providing:- The server is proxying RTP as it has no support for STUN etc.- The NAT is the basic domestic router style, not a full blown firewall requiring port mappings
SimonOn 8/24/06, Andreas Sikkema [EMAIL PROTECTED] wrote:
 Anyone here use the Nokia E61 ? I am looking to invest in a wifi phone and I want to get the best. Is it good as far as reception ? That is of most importance to me. Thanks.I've tried it in the last couple of days. The biggest issue for
me ist that it HAS to be on the same side of a NAT as theserver it talks to (asterisk, ser, etc). If it is on theprivate side of a NAT and the server is on the public side, itdoesn't work. I've read something on the Nokia forums that
Nokia is aware of the problem and it will be solved.My problem is that they want to solve this using STUN etc,while I would prefer they also wouldn't have the softwarecare if it is on the inside of a NAT like most other CPE's
so our platform can take care of things.--Andreas SikkemaBBeyondSoftware EngineerPlaneetbaan 4+31 (0)23 70743422132 HZ Hoofddorp___
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SV: [asterisk-users] E61

[Jon Schøpzinsky]

I also have this phone, and have stumbled in to the same problem.
I just think that it isn't in nokia's interest to change this, as it forces 
consumers to have some sort of local hardware, that (possibly) only the telecom 
provider can give them. This forces the users away from using cheaper services.
Nokia makes a load from the telecom operators around the world, and are not 
interested in pissing them off, by letting their users bypass their price 
structure.

Just my 5 cents.

Jon

-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Andreas Sikkema
Sendt: 24. august 2006 15:24
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: RE: [asterisk-users] E61

 Anyone here use the Nokia E61 ? I am looking to invest in a 
 wifi phone and I want to get the best. Is it good as far as 
 reception ? That is of most importance to me. Thanks.

I've tried it in the last couple of days. The biggest issue for 
me ist that it HAS to be on the same side of a NAT as the 
server it talks to (asterisk, ser, etc). If it is on the 
private side of a NAT and the server is on the public side, it 
doesn't work. I've read something on the Nokia forums that 
Nokia is aware of the problem and it will be solved.

My problem is that they want to solve this using STUN etc, 
while I would prefer they also wouldn't have the software 
care if it is on the inside of a NAT like most other CPE's 
so our platform can take care of things.

-- 
Andreas SikkemaBBeyond
Software EngineerPlaneetbaan 4
+31 (0)23 70743422132 HZ Hoofddorp 
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-- 
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.405 / Virus Database: 268.11.5/426 - Release Date: 23-08-2006
 

-- 
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.405 / Virus Database: 268.11.5/426 - Release Date: 23-08-2006
 
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RE: [asterisk-users] Re: Nokia E60/61/70 and SIP

[Haspers]

I've got them all. It registers correctly with Asterisk, 
and get incoming calls, but it complaints about outgoing calls (Connection 
Error). SIP Debug is giving me: SIP/2.0 407 
Proxy Authentication Required

But 
those settings are the same (Proxy Server/Registrar Server). So what could be 
the problem?


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Simon 
WoodheadSent: donderdag 24 augustus 2006 12:51To: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[asterisk-users] Re: Nokia E60/61/70 and SIP
Hi Haspers,Makes sure you have created an 'Internet tel' 
profile. It doesn't appear to do anything but was vital in getting it working 
for me. The other settings in the how to look sensible.Simon
On 8/24/06, Haspers 
[EMAIL PROTECTED] wrote:
Strange,What 
  settings do you use? I followed this linkhttp://www.newlc.com/Using-SIP-with-Nokia-Series60-and.html 
  but without anyluck. -Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] 
  ] On Behalf Of Benny AmorsenSent: donderdag 24 augustus 2006 
  11:24To: asterisk-users@lists.digium.comSubject: 
  [asterisk-users] Re: Nokia E60/61/70 and SIP  "H" 
  == Haspers[EMAIL PROTECTED] writes:H We 
  are using some E61 and E70's with asterisk. Only problem we haveH at 
  this moment is that we are unable to use a password for the H 
  authentication. I haven't found out yet why this isn't working.H They 
  are working good, but I would like to see some small thingsH changed 
  in future firmware versions (like being able to selectH multiple WLAN 
  points (Access groups) instead of just one.E70 works with passwords 
  here. No trouble.The main issue is that the E70 can't automatically 
  switch to cellular whenthe phone is out of WLAN coverage. It is a bit 
  silly to have to click 
  option-s___--Bandwidth 
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[asterisk-users] About IVR and Oracle (Tim Panton)

[leoarellan]

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[asterisk-users] Monitoring/Listening In (Scott Pinhorne)

[leoarellan]

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RE: [asterisk-users] E61

[Andreas Sikkema]

Simon,

 That is incorrect. It works just fine through NAT providing:
 
 - The server is proxying RTP as it has no support for STUN etc.
 - The NAT is the basic domestic router style, not a full 
 blown firewall requiring port mappings 

Strange, then you must have some other firmware, because I just 
can't get it registered at all, let alone make calls. 

We do have proxies for RTP ;-)

-- 
Andreas SikkemaBBeyond
Software EngineerPlaneetbaan 4
+31 (0)23 70743422132 HZ Hoofddorp 
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Re: [asterisk-users] E61

[Simon Woodhead]

Hi Andreas,I'm on 1.0610.04.04 19-04-06 RM-89WOn 8/24/06, Andreas Sikkema [EMAIL PROTECTED]
 wrote:Simon, That is incorrect. It works just fine through NAT providing:
 - The server is proxying RTP as it has no support for STUN etc. - The NAT is the basic domestic router style, not a full blown firewall requiring port mappingsStrange, then you must have some other firmware, because I just
can't get it registered at all, let alone make calls.We do have proxies for RTP ;-)--Andreas SikkemaBBeyondSoftware EngineerPlaneetbaan 4+31 (0)23 70743422132 HZ Hoofddorp
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Re: [asterisk-users] SpanDSP Error

[Patrick]

On Thu, 2006-08-24 at 09:29 -0400, Christian Jensen wrote:
 I have not found any solution to the problem I am talking about in  
 the archives Steve. I have reinstalled and downgraded from pre26 to  
 pre21 of  SpanDSP 0.0.2 and still to no avail. I have followed all  
 the instructions and ways of fixing this problem and have found none  
 to be a solution.
 -configured with prefix /usr
 -make
 -make install
 -then moved the patch file and the corresponding .c files into the  
 apps directory of my asterisksource
 -Then patched the Makefile and recompiled asterisk. I have done it  
 with about 4 versions so far of which I cannot recall which ones.  
 Pre26 and Pre21 definately
 Anyone?

I don't know which distro you use but the (S)RPMs at
http://www.laimbock.com/asterisk/ have all the spandsp, rxfax and txfax
goodies included (and Unicall too). For Asterisk 1.2.x you need
spandsp-0.0.2pre26. You could check out the patches in the asterisk SRPM
to get an idea how it was done.

Regards,
Patrick

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Re: [asterisk-users] Cisco Router QOS and IAX2

[Bruce Reeves]

Thank you so much. After fighting with a large/extensive QOS policy from Cisco's SDM tool, I used your sample and tweaked it for my needs and everything started working fine.Bruce
On 8/23/06, Rich Adamson [EMAIL PROTECTED] wrote:
Bruce Reeves wrote: I'm needing some pointers from anyone who has been able to get Cisco routers to recognize the iax protocol and perform QOS on it. Or if there is a better way to get my iax traffic prioritized by the router.
You can either match on udp/4569, or, match on TOS header bits. I likeusing the TOS header bits personally as lots of other protocols (eg,dns) will eventually match on udp/4569.For the TOS bits  
v1.2.10, use tos=lowdelay in iax.conf and on thecisco use an access list to match on the tos bits. Something like:access-list 103 permit ip any any dscp cs3access-list 103 permit ip any any dscp efaccess-list 103 permit ip any any tos min-delay= same as tos=lowdelay
access-list 103 permit ip any any tos 12For the TOS bits  svn truck, the tos= settings have changed inasterisk. Look in the supplied documentation (eg, readme's, sampleconfigs) for exactly what is allowed in terms of DiffServ (new term for
TOS basically). You'll find examples that support the above access listitem dscp cs3 and dscp ef.If you're not all that experienced on cisco qos, then the following isan example of a working config that you should be able to translate into
your router config one way or another.class-map match-all voice-rtp match access-group 103class-map match-all www-traffic match access-group 105!policy-map voice-policy class voice-rtp
 priority percent 40 class www-trafficbandwidth percent 30 class class-defaultfair-queue!interface Dialer0bandwidth 555snip, my specific interface config statements
service-policy output voice-policy!access-list 103 permit ip any any dscp cs3access-list 103 permit ip any any dscp efaccess-list 103 permit ip any any tos min-delayaccess-list 103 permit ip any any tos 12
access-list 105 permit tcp any eq www anyThe above config provides low-latency priority to voice-rtp, thenprovides an additional qos piece to ensure www-traffic is givenbandwidth before all of the class-default traffic. In other words,
voice-rtp traffic will always get 40% of the bandwidth (eg, 40% ofbandwidth=555 above) if voice traffic is present. If voice trafficisn't present, that bandwidth can be used by other qos sections or by
the default class. Same with www-traffic after the router deals withvoice-rtp traffic. The default class always gets what bandwidth is leftover (or all bandwidth if there is no voice-rtp or www-traffic).
To troubleshoot the above, do a show access-list 103 from the CLI (onthe router) and watch for matching packets in each access list line.Once you've structured the access list to truly match asterisk traffic,
then do a show policy-map interface dialer0 to display how the overallqos structure is functioning.Note that cisco didn't get real serious about IOS qos until v12.2 oftheir IOS code. In v12.2
 (and later versions of IOS) there has been asignificant amount of work to bring all of their products into industrystandard implementations / conformance / expectations. If you want toget real serious with cisco's qos stuff, purchase the book End-to-end
QoS Network Design and read the 700+ pages devoted to the subject. Itis an excellent book with lots of examples, etc. The book (and actualpractice) suggests IOS v12.3 has more QoS funtionality then v12.2
, andv12.4 has more then v12.3. (The authors of the book back that statementup 100% as well, and they are cisco employees.)In the above config, the bandwidth=555 statement is very important. It
should represent the actual outgoing bandwidth for whatever interfaceyou are using and not the theoretical max that someone said you should get.Also note that for relatively slow speed interfaces (eg, most dsl's),
the outgoing bandwidth is rather slow. If you calculate how much time isconsumed sending a non-voice 1500-byte packet, the time is likely to bemore then the 20 millisecond interval for sip/iax traffic. If that is
your case, then you may need to forcibly reduce the MTU size of packetsoriginating from other non-voice workstations/servers. The later ciscoIOS versions have a parameter to do that if you can't do it via the
workstation/server configuration parameters. If memory serves correctly,that parameter appeared around v12.4 of their IOS.One last item... all of the above deals only with outgoing traffic.You would need to talk to your ISP about QoS for your incoming traffic,
and most of the local ISP's don't have a clue. Increasingly, some of thelarger backbone isp's are beginning to understand QoS and some haveactually implemented something. However, those isp's are heading towards
providing QoS as a value-add chargeable service (as in MPLS, etc).R.___--Bandwidth and Colocation provided by Easynews.com
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Re: [asterisk-users] Re: Nokia E60/61/70 and SIP

[Simon Woodhead]

Hi,Yes, the proxy and registrar settings are identical in my set-up. Getting registered is the hard part but you've done that. I'd be looking at the Asterisk end of things rather than the E61 to see why that authentication issue is arising.
WOn 8/24/06, Haspers [EMAIL PROTECTED] wrote:





I've got them all. It registers correctly with Asterisk, 
and get incoming calls, but it complaints about outgoing calls (Connection 
Error). SIP Debug is giving me: SIP/2.0 407 
Proxy Authentication Required

But 
those settings are the same (Proxy Server/Registrar Server). So what could be 
the problem?


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED]] On Behalf Of Simon 
WoodheadSent: donderdag 24 augustus 2006 12:51To: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[asterisk-users] Re: Nokia E60/61/70 and SIP
Hi Haspers,Makes sure you have created an 'Internet tel' 
profile. It doesn't appear to do anything but was vital in getting it working 
for me. The other settings in the how to look sensible.Simon
On 8/24/06, Haspers 
[EMAIL PROTECTED] wrote:
Strange,What 
  settings do you use? I followed this linkhttp://www.newlc.com/Using-SIP-with-Nokia-Series60-and.html
 
  but without anyluck. -Original Message-From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] 
  ] On Behalf Of Benny AmorsenSent: donderdag 24 augustus 2006 
  11:24To: asterisk-users@lists.digium.comSubject: 
  [asterisk-users] Re: Nokia E60/61/70 and SIP  H 
  == Haspers[EMAIL PROTECTED] writes:H We 
  are using some E61 and E70's with asterisk. Only problem we haveH at 
  this moment is that we are unable to use a password for the H 
  authentication. I haven't found out yet why this isn't working.H They 
  are working good, but I would like to see some small thingsH changed 
  in future firmware versions (like being able to selectH multiple WLAN 
  points (Access groups) instead of just one.E70 works with passwords 
  here. No trouble.The main issue is that the E70 can't automatically 
  switch to cellular whenthe phone is out of WLAN coverage. It is a bit 
  silly to have to click 
  option-s___--Bandwidth 
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  visit:  http://lists.digium.com/mailman/listinfo/asterisk-users
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  visit: http://lists.digium.com/mailman/listinfo/asterisk-users 
  

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Re: [asterisk-users] E61

[Thomas Artner]

Hi!

I can't find the link anymore where it  was a statement from the nokia
support that they are working on a STUN implementation.
A firmwareupdate (with STUN support) will be available in fall 2006.

tom


Andreas Sikkema wrote:
 Anyone here use the Nokia E61 ? I am looking to invest in a 
 wifi phone and I want to get the best. Is it good as far as 
 reception ? That is of most importance to me. Thanks.
 
 I've tried it in the last couple of days. The biggest issue for 
 me ist that it HAS to be on the same side of a NAT as the 
 server it talks to (asterisk, ser, etc). If it is on the 
 private side of a NAT and the server is on the public side, it 
 doesn't work. I've read something on the Nokia forums that 
 Nokia is aware of the problem and it will be solved.
 
 My problem is that they want to solve this using STUN etc, 
 while I would prefer they also wouldn't have the software 
 care if it is on the inside of a NAT like most other CPE's 
 so our platform can take care of things.
 

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Re: [asterisk-users] Active Directory Listing Feauture

[Andrew Latham]

Are you asking about the LDAP stuff or voicemail directory?



On 8/24/06, Mohamed A. Gombolaty [EMAIL PROTECTED] wrote:

 Dear All,

I am currently very stumped on the subject of Active Directory listing, as I
am unable to find any documents regarding this feature thus I am unable to
configure it or know how to use it. Does anyone have any useful info or
documents regarding this feature in terms of how to or guides I will be very
much thankful.

  --
Thx
MAG




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--
---
Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
[EMAIL PROTECTED] - [EMAIL PROTECTED]
If any of the above are down we have bigger problems than my email!
Hind sight is most always 20/20 or better.
---
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Re: [asterisk-users] E61

[Thomas Artner]

here is the link (german):
http://www.my-s60.com/de/news/news/newswann_kann_die_e_serie_nat_traversal/back/105/cHash/bac3645f5e/index.html

STUN will come in fall 2006
TURN and ICE in 2007


Thomas Artner wrote:
 Hi!
 
 I can't find the link anymore where it  was a statement from the nokia
 support that they are working on a STUN implementation.
 A firmwareupdate (with STUN support) will be available in fall 2006.
 
 tom
 
 
 Andreas Sikkema wrote:
 Anyone here use the Nokia E61 ? I am looking to invest in a 
 wifi phone and I want to get the best. Is it good as far as 
 reception ? That is of most importance to me. Thanks.
 I've tried it in the last couple of days. The biggest issue for 
 me ist that it HAS to be on the same side of a NAT as the 
 server it talks to (asterisk, ser, etc). If it is on the 
 private side of a NAT and the server is on the public side, it 
 doesn't work. I've read something on the Nokia forums that 
 Nokia is aware of the problem and it will be solved.

 My problem is that they want to solve this using STUN etc, 
 while I would prefer they also wouldn't have the software 
 care if it is on the inside of a NAT like most other CPE's 
 so our platform can take care of things.

 
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Re: [asterisk-users] Cisco Router QOS and IAX2

[Rich Adamson]

The majority of the sample qos policies seem to be based on either five 
or seven qos queues, and most folks don't need all of that. What I've 
shown as a sample only has three queues; one for voip, one for my 
outbound web traffic, and the default queue that everything else falls 
into.


You can actually remove the sections relating to web traffic if you 
don't have a production web server contending for outbound traffic, 
making it a two-queue policy.


R.

Bruce Reeves wrote:
Thank you so much. After fighting with a large/extensive QOS policy from 
Cisco's SDM tool, I used your sample and tweaked it for my needs and 
everything started working fine.


Bruce

On 8/23/06, *Rich Adamson* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


Bruce Reeves wrote:
  I'm needing some pointers from anyone who has been able to get Cisco
  routers to recognize the iax protocol and perform QOS on it. Or
if there
  is a better way to get my iax traffic prioritized by the router.
 

You can either match on udp/4569, or, match on TOS header bits. I like
using the TOS header bits personally as lots of other protocols (eg,
dns) will eventually match on udp/4569.

For the TOS bits  v1.2.10, use tos=lowdelay in iax.conf and on the
cisco use an access list to match on the tos bits. Something like:
access-list 103 permit ip any any dscp cs3
access-list 103 permit ip any any dscp ef
access-list 103 permit ip any any tos min-delay  = same as
tos=lowdelay
access-list 103 permit ip any any tos 12

For the TOS bits  svn truck, the tos= settings have changed in
asterisk. Look in the supplied documentation (eg, readme's, sample
configs) for exactly what is allowed in terms of DiffServ (new term for
TOS basically). You'll find examples that support the above access list
item dscp cs3 and dscp ef.

If you're not all that experienced on cisco qos, then the following is
an example of a working config that you should be able to translate
into
your router config one way or another.

class-map match-all voice-rtp
   match access-group 103
class-map match-all www-traffic
   match access-group 105
!
policy-map voice-policy
   class voice-rtp
 priority percent 40
   class www-traffic
bandwidth percent 30
   class class-default
fair-queue
!
interface Dialer0
  bandwidth 555
  snip, my specific interface config statements
  service-policy output voice-policy
!
access-list 103 permit ip any any dscp cs3
access-list 103 permit ip any any dscp ef
access-list 103 permit ip any any tos min-delay
access-list 103 permit ip any any tos 12
access-list 105 permit tcp any eq www any

The above config provides low-latency priority to voice-rtp, then
provides an additional qos piece to ensure www-traffic is given
bandwidth before all of the class-default traffic. In other words,
voice-rtp traffic will always get 40% of the bandwidth (eg, 40% of
bandwidth=555 above) if voice traffic is present. If voice traffic
isn't present, that bandwidth can be used by other qos sections or by
the default class. Same with www-traffic after the router deals with
voice-rtp traffic. The default class always gets what bandwidth is left
over (or all bandwidth if there is no voice-rtp or www-traffic).

To troubleshoot the above, do a show access-list 103 from the CLI (on
the router) and watch for matching packets in each access list line.
Once you've structured the access list to truly match asterisk traffic,
then do a show policy-map interface dialer0 to display how the overall
qos structure is functioning.

Note that cisco didn't get real serious about IOS qos until v12.2 of
their IOS code. In v12.2 (and later versions of IOS) there has been a
significant amount of work to bring all of their products into industry
standard implementations / conformance / expectations. If you want to
get real serious with cisco's qos stuff, purchase the book End-to-end
QoS Network Design and read the 700+ pages devoted to the subject. It
is an excellent book with lots of examples, etc. The book (and actual
practice) suggests IOS v12.3 has more QoS funtionality then v12.2 , and
v12.4 has more then v12.3. (The authors of the book back that statement
up 100% as well, and they are cisco employees.)

In the above config, the bandwidth=555 statement is very
important. It
should represent the actual outgoing bandwidth for whatever interface
you are using and not the theoretical max that someone said you
should get.

Also note that for relatively slow speed interfaces (eg, most dsl's),
the outgoing bandwidth is rather slow. If you calculate how much time is
consumed sending a non-voice 1500-byte packet, the time is likely to be
more then the 20 millisecond interval for 

Re: [asterisk-users] SpanDSP Error

[Steve Underwood]

Christian Jensen wrote:

I have not found any solution to the problem I am talking about in  
the archives Steve. I have reinstalled and downgraded from pre26 to  
pre21 of  SpanDSP 0.0.2 and still to no avail. I have followed all  
the instructions and ways of fixing this problem and have found none  
to be a solution.

-configured with prefix /usr
-make
-make install
-then moved the patch file and the corresponding .c files into the  
apps directory of my asterisksource
-Then patched the Makefile and recompiled asterisk. I have done it  
with about 4 versions so far of which I cannot recall which ones.  
Pre26 and Pre21 definately

Anyone?


If you look, you will find many occurrences of the answer. Remove 
spandsp 0.0.3 from your system. It is there, and it is causing conflicts.


Steve

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RE: [asterisk-users] SSH connection hangs on logout?

[Rushowr]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Tzafrir Cohen
 Sent: Thursday, August 24, 2006 2:32 AM
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] SSH connection hangs on logout?
 
 On Wed, Aug 23, 2006 at 11:03:23PM -0700, Steve Edwards wrote:
  On Thu, 24 Aug 2006, Jeremy McNamara wrote:
  
  Rushowr wrote:
  Hey all, I have an interesting issue that just recently 
 started when 
  I grabbed a copy of the trunk about a week ago and 
 compiled it. Ever 
  since that compile, if I start Asterisk (disconnected terminal, 
  using safe_asterisk to launch) and then continue on about my work 
  with it, when I disconnect my SSH terminal (using latest 
 version of 
  PuTTY) the session no longer closes it just hangs. I've 
 even changed 
  the Putty setting to close the window even on unclean exit but it 
  still hangs the connection... I had something similar once with 
  Zabbix a while back, but never Asterisk.
  
  Anyone else experience this?
  
  Start asterisk using  safe_asterisk or via asterisk -f
  
  I prefer the safe_asterisk shell script, since if asterisk seg 
  faults, there is a good chance asterisk will get 
 automatically restarted.
  
  Jeremy McNamara
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  You may need to redirect stdin, stdout, stderr like:
  
  run_asterisk\
  0/dev/null\
  1/dev/null\
  2/dev/null\
  
  
 
 In other words: 
 
 A plain 'asterisk' (without '-c' and such) that daemonizes 
 and does exactly that for you, among others.
 
 Asterisk is a daemon, rather than an interactive program. 
 Thus its handling for SIGHUP is to re-read configuration 
 rather than detach from the terminal.
 
 -- 
 Tzafrir Cohen sip:[EMAIL PROTECTED]
 icq#16849755  iax:[EMAIL PROTECTED]
 +972-50-7952406  jabber:[EMAIL PROTECTED]
 [EMAIL PROTECTED] http://www.xorcom.com
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Shoots out own flames before emailing

Gents, asstated in my email, I am using safe_asterisk. Additionally, even
when I started asterisk by hand, it was always forked off of my tty.
However, even if I DID have it connected to my tty, I'd have to issue stop
now before getting to the command prompt and being able to issue logout
to bash. 

Try this sometime gents, you'll see what I mean...issue a ! From the
*CLI...then type logout...You'll be told that you're not in a login shell
and to use exit.


Wow..
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.5 (MingW32)
Comment: ENCRYPTED WITH GPG

iD8DBQFE7cEuwWoA8HY7JXYRAqtoAJwNX8/L7OFuXvTPobOvJ8cH0Iei9QCaApf0
S4BH1uc4ZxWxei0gRy+qKy0=
=6PsA
-END PGP SIGNATURE-



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[asterisk-users] need help with error code

[Michael Sampson]

Can anyone give me some insight as to what this message means from 
/var/log/asterisk/full


Aug 24 10:32:02 DEBUG[3016] chan_zap.c: Write returned -1 (Resource 
temporarily unavailable) on channel 1
Aug 24 10:32:02 DEBUG[3016] chan_zap.c: Write returned -1 (Resource 
temporarily unavailable) on channel 1
Aug 24 10:32:02 DEBUG[3016] chan_zap.c: Write returned -1 (Resource 
temporarily unavailable) on channel 1
Aug 24 10:32:03 DEBUG[3016] chan_zap.c: Write returned -1 (Resource 
temporarily unavailable) on channel 1
Aug 24 10:32:03 DEBUG[3016] chan_zap.c: Write returned -1 (Resource 
temporarily unavailable) on channel 1


I get them during a call that is from a sip phone out a zap channel. The 
asterisk dials out to the PSTN through another pbx system. I am having a 
problem where those calls get disconnected after about 10 mins and was 
wondering if that message had anything to do with it.


--
Michael Sampson
Information Systems Manager
Customer Contact Services
[EMAIL PROTECTED]
952-936-4000

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Re: [asterisk-users] Realtime and hints

[Dovid Bender]

Doug I have
Exten = 10,hint,SIP/11010
and in mysql I have
exten = 10,1,Dial(SIP/11010)

- Original Message - 
From: Douglas Garstang [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Monday, August 21, 2006 3:37 PM
Subject: [asterisk-users] Realtime and hints


Can realtime be used with hints? How would you get the following into the 
database given that the priority column is numeric, and that there is no 
application for the first entry?


exten = 2944006,hint,SIP/2944006
exten = 2944006,1,Dial(SIP/2944006)

Every time I touch realtime I hit obstacles. How are others getting around 
this limitation?


Doug.

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Re: [asterisk-users] Realtime Extensions -- Comments?

[Dovid Bender]

Maybe he is tryin to make it work. This is much better than the old Doug. 
Also if he needs features that currently dont exist maybe some one will 
create it and then we will all benefit from it :) .
- Original Message - 
From: Brian Capouch [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Thursday, August 24, 2006 12:43 AM
Subject: Re: [asterisk-users] Realtime Extensions -- Comments?



Douglas Garstang wrote:



It doesn't matter where you turn in Asterisk, there's gotcha's. For 
example, you can't put the hint stuff into realtime, and there's no 
inherint way to comment extensions.


It doesn't seem like Asterisk is good enough for you Doug.

Switch to one of the competitors' products.

B.

--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.

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[asterisk-users] SendText Queue Notification

[John D. Coleman]

I was wondering if anyone was able to execute custom commands on a
channel once a caller connects to an agent after being in a queue.  The
reason I ask, is because I would like to use SendText to send a message
to the agent receiving the call to let the agent know how many calls are
waiting in the queue.  I tried using ChanSpy, but then SendText will
send messages only to and from the caller who initiated the ChanSpy.

One way I could get around this is if I found out how to use SendText
from the commandline, like smsq. I'm pretty sure that's not possible
because of the nature of SIP MESSAGE but I figured I'd ask.

Thanks,

John Coleman
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[asterisk-users] originate from group + dialplan

[dmb]

Hello,
is possible execute a dialplan before make a call from Zap/g0/dnis? How can
i do that?

Thank you

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[asterisk-users] Wellgate 3804a

[Ronald Wiplinger]

I want that each call from PSTN goes to Asterisk to the context for this 
line. Within this context can be a menu or a dial command, ...

As more I read, as more I get confused, ... and each try is not working!


My sip.conf:

[WG88621001] 
type=friend 
defaultip=192.168.250.244

insecure=very
context=incoming_WG
dtmfmode=rfc2833
[EMAIL PROTECTED]
language=en
nat=yes
auth=md5
host=dynamic
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=g726
allow=g729
username=88621001
fromuser=88621001   
secret= 
qualify=yes

canreinvite=no


extensions.conf
[incoming_WG]
exten = s,1,NoOp(*** I am here now ***)



Wellgate settings:

Network Interface: IP address of the device 192.168.250.244

Sip Config: Mode Proxy
   Primary Proxy IP address:  192.168.250.20
   Line 1 Number:88621001
  
Security Config

   Line1 Account:  WG8862001
   Line 1 password:   (secret from the asterisk setting)

Line configuration
   Line 1 (LINE)   Type: FXO   Hunting Group: 1   HotLine: 601   
Registration: Not Registered   Status: Ready


System Configuration
   Keypad type:   rfc2833

Routing Table
   Index: IP Default Destination: FXO  E.164: x
   Index: FXO Destination: IP Default  E.164: x



*CLI sip show peers like ^WG
Name/username  HostDyn Nat ACL Port Status   
WG88621001/88621001(Unspecified)D   N  0UNKNOWN  
1 sip peers [0 online , 1 offline]




Calls from PSTN comes to the IVR asking for the extension number and 
than nothing happens. Asterisk shows nothing either.


Can somebody enlighten me:
1. Do I need to have a register statement in sip.conf?
(I tried register = 88621001:secret-from-above   ; Wellgate GW.3801-Line-1)

2. where to turn off the IVR?

3. Do I use the right  name, user name, line account, line 

4. Hotline. Why, how, which number??


bye

Ronald


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Re: [asterisk-users] No CLID from PSTN using X100P FXO Card

[William Moore]

However, reinstalled the box from ground up and installed 1.2.10 and now
CLID isn't working at all.   The PSTN line is still transmitting it, as
I've plugged in my Uniden cordless with CLID and it shows up fine on
there, but getting absolutely nothing inside the ${CALLERIDNUM} and
${CALLERIDNAME} variables.


In 1.2.10, these variables have been changed to a single function.
The new way to access those would be ${CALLERID(name)} and
${CALLERID(num)}.
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Re: [asterisk-users] SendText Queue Notification

[Brodie Macleod]

I know this isn't answering your question, but what I did for queue 
notification was use softkeys on the phones that call a PHP script on the * 
box that'll output XML for the phone to parse and display the queue stats on 
demand. Of course your phone would need to have an XML parser or some other 
type of minibrowser.  For sending SIP messages to my Snom phones I use Sipsak 
to display agent login info and their associated queue(s) so that it's easy 
for agents to know what their status is.

-Brodie

On Thursday 24 August 2006 10:33 am, John D. Coleman wrote:
 I was wondering if anyone was able to execute custom commands on a
 channel once a caller connects to an agent after being in a queue.  The
 reason I ask, is because I would like to use SendText to send a message
 to the agent receiving the call to let the agent know how many calls are
 waiting in the queue.  I tried using ChanSpy, but then SendText will
 send messages only to and from the caller who initiated the ChanSpy.

 One way I could get around this is if I found out how to use SendText
 from the commandline, like smsq. I'm pretty sure that's not possible
 because of the nature of SIP MESSAGE but I figured I'd ask.

 Thanks,

 John Coleman
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[asterisk-users] Re: channel variable

[Tony Mountifield]

In article [EMAIL PROTECTED],
unplug [EMAIL PROTECTED] wrote:
 Tony Mountifield wrote:
 
  Try using Set(__testmode=1), but still refer to it as ${testmode}.
 
  The __ tells Asterisk to propagate the variable to created channels.
 
 Do you mean it is a global variable instead of channel variable?

No, it is a channel variable. The __ on the beginning means it will be
inherited by any channels created from the channel it was set on.

See the section called Inheritance of Channel Variables on
http://www.voip-info.org/wiki/view/Asterisk+variables

In fact you will find that there is a LOT of useful info on that site.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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RE: [asterisk-users] Realtime and hints

[Douglas Garstang]

But... you need _both_ in your dialplan.

My extensions.conf has:

exten = 2944054,hint,  SIP/2944054
exten = 2944054,1, Dial(SIP/2944054)

ie two lines for the hint.


 -Original Message-
 From: Dovid Bender [mailto:[EMAIL PROTECTED]
 Sent: Thursday, August 24, 2006 9:32 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Realtime and hints
 
 
 Doug I have
 Exten = 10,hint,SIP/11010
 and in mysql I have
 exten = 10,1,Dial(SIP/11010)
 
 - Original Message - 
 From: Douglas Garstang [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Monday, August 21, 2006 3:37 PM
 Subject: [asterisk-users] Realtime and hints
 
 
 Can realtime be used with hints? How would you get the 
 following into the 
 database given that the priority column is numeric, and that 
 there is no 
 application for the first entry?
 
 exten = 2944006,hint,SIP/2944006
 exten = 2944006,1,Dial(SIP/2944006)
 
 Every time I touch realtime I hit obstacles. How are others 
 getting around 
 this limitation?
 
 Doug.
 
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RE: [asterisk-users] Realtime and hints

[Aaron Daniel]

That's what he was gettin at.  Take the second line out, and put the
first priority in the database.

On Thu, 2006-08-24 at 10:17 -0600, Douglas Garstang wrote:
 But... you need _both_ in your dialplan.
 
 My extensions.conf has:
 
 exten = 2944054,hint,  SIP/2944054
 exten = 2944054,1, Dial(SIP/2944054)
 
 ie two lines for the hint.
 
 
  -Original Message-
  From: Dovid Bender [mailto:[EMAIL PROTECTED]
  Sent: Thursday, August 24, 2006 9:32 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] Realtime and hints
  
  
  Doug I have
  Exten = 10,hint,SIP/11010
  and in mysql I have
  exten = 10,1,Dial(SIP/11010)
  
  - Original Message - 
  From: Douglas Garstang [EMAIL PROTECTED]
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  asterisk-users@lists.digium.com
  Sent: Monday, August 21, 2006 3:37 PM
  Subject: [asterisk-users] Realtime and hints
  
  
  Can realtime be used with hints? How would you get the 
  following into the 
  database given that the priority column is numeric, and that 
  there is no 
  application for the first entry?
  
  exten = 2944006,hint,SIP/2944006
  exten = 2944006,1,Dial(SIP/2944006)
  
  Every time I touch realtime I hit obstacles. How are others 
  getting around 
  this limitation?
  
  Doug.
  
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-- 
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198

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[asterisk-users] No outbound with A2Billing

[Luciano Moreira]

List members,

When I dial to a PSTN number, the A2Billing script does all the tasks,
until it shutdown without make the dailout by sip trunk set.
Lasts outputs fro the a2billing.php debug are:
  a2billing.php|2: RESFINDRATE:: 0
  a2billing.php|2: UPDATE cc_card SET inuse=inuse-1 WHERE username='5033845534'

Sip trunk is registered and working. All setups in A2Billing db seams ok.

There is any a2billing guru to help me?

Below, is the complete script output.

Thank you in advance.

Luc Moreira
__
Logic Telecom
Fortaleza, Brasil
---
-- Accepting AUTHENTICATED call from 201.49.16.125:
requested format = g723,
requested prefs = (),
actual format = g729,
host prefs = (g729|g723|gsm|ulaw),
priority = mine
-- Executing Answer(IAX2/1010-15, ) in new stack
-- Executing Wait(IAX2/1010-15, 0) in new stack
-- Executing DeadAGI(IAX2/1010-15, a2billing.php|2) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
  a2billing.php|2: IDCONFIG : 2
  a2billing.php|2:
  a2billing.php|2: A2Billing AGI internal configuration:
  a2billing.php|2: Array
  a2billing.php|2: (
  a2billing.php|2: [debug] = 3
  a2billing.php|2: [answer_call] = 1
  a2billing.php|2: [logger_enable] = 1
  a2billing.php|2: [log_file] = /tmp/a2billing.log
  a2billing.php|2: [say_goodbye] =
  a2billing.php|2: [play_menulanguage] =
  a2billing.php|2: [force_language] = br
  a2billing.php|2: [len_cardnumber] = 10
  a2billing.php|2: [len_aliasnumber] = 5
  a2billing.php|2: [len_voucher] = 15
  a2billing.php|2: [min_credit_2call] = 0
  a2billing.php|2: [min_duration_2bill] = 20
  a2billing.php|2: [notenoughcredit_cardnumber] = 1
  a2billing.php|2: [notenoughcredit_assign_newcardnumber_cid] = 1
  a2billing.php|2: [use_dnid] = 1
  a2billing.php|2: [no_auth_dnid] = Array
  a2billing.php|2: (
  a2billing.php|2: [0] =
  a2billing.php|2: )
  a2billing.php|2:
  a2billing.php|2: [number_try] = 1
  a2billing.php|2: [say_balance_after_auth] =
  a2billing.php|2: [say_balance_after_call] =
  a2billing.php|2: [say_rateinitial] =
  a2billing.php|2: [say_timetocall] =
  a2billing.php|2: [auto_setcallerid] = 1
  a2billing.php|2: [force_callerid] =
  a2billing.php|2: [cid_sanitize] =
  a2billing.php|2: [cid_enable] = 1
  a2billing.php|2: [cid_askpincode_ifnot_callerid] = 1
  a2billing.php|2: [cid_auto_create_card] = 1
  a2billing.php|2: [cid_auto_assign_card_to_cid] = 1
  a2billing.php|2: [cid_auto_create_card_typepaid] = POSTPAY
  a2billing.php|2: [cid_auto_create_card_credit] = 0
  a2billing.php|2: [cid_auto_create_card_credit_limit] = 100
  a2billing.php|2: [cid_auto_create_card_tariffgroup] = 1
  a2billing.php|2: [callerid_authentication_over_cardnumber] =
  a2billing.php|2: [sip_iax_friends] =
  a2billing.php|2: [sip_iax_pstn_direct_call_prefix] = 9
  a2billing.php|2: [sip_iax_pstn_direct_call] =
  a2billing.php|2: [dialcommand_param] = |60|HL(%timeout%:61000:3,Ttr)
  a2billing.php|2: [dialcommand_param_sipiax_friend] = 
|60|HL(360:61000:3,Ttr)
  a2billing.php|2: [switchdialcommand] = 1
  a2billing.php|2: [maxtime_tocall_negatif_free_route] = 3600
  a2billing.php|2: [send_reminder] = 1
  a2billing.php|2: [record_call] =
  a2billing.php|2: [monitor_formatfile] = gsm
  a2billing.php|2: [base_currency] = usd
  a2billing.php|2: [agi_force_currency] = usd
  a2billing.php|2: [currency_association] = Array
  a2billing.php|2: (
  a2billing.php|2: [0] = usd:prepaid-dollar
  a2billing.php|2: [1] = mxn:pesos
  a2billing.php|2: [2] = eur:euro
  a2billing.php|2: [3] = all:credit
  a2billing.php|2: [4] = brl:credit
  a2billing.php|2: )
  a2billing.php|2:
  a2billing.php|2: [file_conf_enter_destination] = prepaid-enter-dest
  a2billing.php|2: [file_conf_enter_menulang] = prepaid-menulang2
  a2billing.php|2: [setlanguage_deprecate] = 1
  a2billing.php|2: [currency_association_internal] = Array
  a2billing.php|2: (
  a2billing.php|2: [usd] = prepaid-dollar
  a2billing.php|2: [mxn] = pesos
  a2billing.php|2: [eur] = euro
  a2billing.php|2: [all] = credit
  a2billing.php|2: [brl] = credit
  a2billing.php|2: )
  a2billing.php|2:
  a2billing.php|2: )
  a2billing.php|2:
  a2billing.php|2: AGI Request:
  a2billing.php|2: Array
  a2billing.php|2: (
  a2billing.php|2: [agi_request] = a2billing.php
  a2billing.php|2: [agi_channel] = IAX2/1010-15
  a2billing.php|2: [agi_language] = br
  a2billing.php|2: [agi_type] = IAX2
  a2billing.php|2: [agi_uniqueid] = 1156436221.21
  a2billing.php|2: [agi_callerid] = 1010
  a2billing.php|2: 

Re: [asterisk-users] E61

[Brodie Macleod]

Using STUN isn't a solution to NAT either, as it won't work with symmetrical 
NAT, which is very common (or for at least to partially use symmetrical).

I'll be interested to see how the Paragon Wifi phone fares out when it starts 
making an appearance in the US.

-Brodie

On Thursday 24 August 2006 08:23 am, Andreas Sikkema wrote:
  Anyone here use the Nokia E61 ? I am looking to invest in a
  wifi phone and I want to get the best. Is it good as far as
  reception ? That is of most importance to me. Thanks.

 I've tried it in the last couple of days. The biggest issue for
 me ist that it HAS to be on the same side of a NAT as the
 server it talks to (asterisk, ser, etc). If it is on the
 private side of a NAT and the server is on the public side, it
 doesn't work. I've read something on the Nokia forums that
 Nokia is aware of the problem and it will be solved.

 My problem is that they want to solve this using STUN etc,
 while I would prefer they also wouldn't have the software
 care if it is on the inside of a NAT like most other CPE's
 so our platform can take care of things.
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RE: [asterisk-users] Realtime and hints

[Douglas Garstang]

I don't see how that helps. If you have a portion of the hint still in 
extensions.conf, then what use is the database?

 -Original Message-
 From: Aaron Daniel [mailto:[EMAIL PROTECTED]
 Sent: Thursday, August 24, 2006 10:24 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [asterisk-users] Realtime and hints
 
 
 That's what he was gettin at.  Take the second line out, and put the
 first priority in the database.
 
 On Thu, 2006-08-24 at 10:17 -0600, Douglas Garstang wrote:
  But... you need _both_ in your dialplan.
  
  My extensions.conf has:
  
  exten = 2944054,hint,  SIP/2944054
  exten = 2944054,1, Dial(SIP/2944054)
  
  ie two lines for the hint.
  
  
   -Original Message-
   From: Dovid Bender [mailto:[EMAIL PROTECTED]
   Sent: Thursday, August 24, 2006 9:32 AM
   To: Asterisk Users Mailing List - Non-Commercial Discussion
   Subject: Re: [asterisk-users] Realtime and hints
   
   
   Doug I have
   Exten = 10,hint,SIP/11010
   and in mysql I have
   exten = 10,1,Dial(SIP/11010)
   
   - Original Message - 
   From: Douglas Garstang [EMAIL PROTECTED]
   To: Asterisk Users Mailing List - Non-Commercial Discussion 
   asterisk-users@lists.digium.com
   Sent: Monday, August 21, 2006 3:37 PM
   Subject: [asterisk-users] Realtime and hints
   
   
   Can realtime be used with hints? How would you get the 
   following into the 
   database given that the priority column is numeric, and that 
   there is no 
   application for the first entry?
   
   exten = 2944006,hint,SIP/2944006
   exten = 2944006,1,Dial(SIP/2944006)
   
   Every time I touch realtime I hit obstacles. How are others 
   getting around 
   this limitation?
   
   Doug.
   
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 -- 
 Aaron Daniel
 Computer Systems Technician
 Sam Houston State University
 [EMAIL PROTECTED]
 (936) 294-4198
 
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Re: [asterisk-users] Annoying Bristuff

[Andrew Nowrot]

HiCan anyone confirm a working asterisk 1.2 from bristuff with 1 port PCI, hfc-s based ISDN card (zaphfc driver). If so, could you send your configuration. I mean OS (linux distribution) type, kernel version.
Thanks in advanceCheersAndrew
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[asterisk-users] Asterisk t38passthrough

[Ricardo Carvalho]

Hi,

I've installed Asterisk t38passthrough branch and I'm using one 
Grandstream ATA to connect Asterisk to a Fax machine. Every time I send 
a fax, it gets sent using codec G711, and never T.38. I added the 
following parameters in the [general] section as well as in device 
configurations:


t38pt_udptl = yes
t38pt_rtp = yes
t38pt_tcp = yes


I think that's the only thing that is needed to do to enable T.38 pass 
through...

Why does Asterisk keeps sending in G711? Any help?

Regards,

Ricardo.
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[asterisk-users] Modems dialing over sangoma a104d

[Sean Cook]

I have a sangoma 104d that is our main pbx now( legacy system died ).  I
have replaced every phone in the building and things are going very well.
We have fax working well and calls are routing properly...  All is well...

Except for our support modems... we have support people that dial out with
modems across our PRI's.  These modems are attached to an Adtran 750 with 24
FXS's.  I have disabled echo cancelation on the T1 that is connected to the
Adtran but negotiation is still really rough.  I am bridging across the same
card and it isn't doing very well... has anyone done this with reasonably
successful results?  I am not looking for 56K I am looking for around 9600
to 14.4..

Thanks,

Sean
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[asterisk-users] Call Parking Ring Back (Snoms)

[J. Oquendo]

Quick question maybe someone can point me in the right direction...

Caller -- Receptionist -- ParksCall
Receptionist makes announcement for individual to pick up parked call. 
No one picks up so it rings back to receptionist within a minute and a 
half. Is there any way to change the ringer for a parked call coming 
back since their call wasn't answered?


--

J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743
sil . infiltrated @ net http://www.infiltrated.net 


The happiness of society is the end of government.
John Adams

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[asterisk-users] Quiet on the list today?

[Rushowr]

Just gotta check, I've never seen a complete day with no posts


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[asterisk-users] hotel teledex integration anyone?

[Curt Shaffer]

All,



I am looking at taking on a project for a hotel that is
using Teledex systems. I see that they have a SIP based phone and the information
says that there is some CMS server part that appears to be the brains behind
the device. My questions are; has anyone out there used this type of system
before? Have you integrated it with Asterisk with good success? Any helpful
hints when moving into this market? Has any found a comparable CMS (maybe open
source) that can be used for this industry? I have been watching the list for
these types of posts and I have seen some hotel posts and eagerly read them but
have not seen any recommendation of how to best go towards this type of
project.



Thanks



Curt






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Re: [asterisk-users] Modems dialing over sangoma a104d

[Rich Adamson]

Sean Cook wrote:

I have a sangoma 104d that is our main pbx now( legacy system died ).  I
have replaced every phone in the building and things are going very well.
We have fax working well and calls are routing properly...  All is well...

Except for our support modems... we have support people that dial out with
modems across our PRI's.  These modems are attached to an Adtran 750 with 24
FXS's.  I have disabled echo cancelation on the T1 that is connected to the
Adtran but negotiation is still really rough.  I am bridging across the same
card and it isn't doing very well... has anyone done this with reasonably
successful results?  I am not looking for 56K I am looking for around 9600
to 14.4..


Can we assume that you've got the correct timing parameters set on the 
104d?  (eg, are you sync'ing your 104d from the telco?)


If not, get that corrected first as it makes a major difference with 
modem calls.


The echo cancellation disabling should be automatic I believe, so would 
not expect turning it on/off to have much of an impact.


I've done this with the sangoma's analog card (a200d) and modem calls 
work very well. I didn't actually check the speed, but felt like 14.4 or 
better.


One of the stated test suites (by sangoma employees) is to validate all 
hardware and driver designs by testing with modems since that really is 
one of the most critical non-test-equipment tests that can be done. So, 
I've got to believe it works; just need to identify the missing link.


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Re: [asterisk-users] RE: [asterisk-dev] Phone status

[Mir]

What do you mean?

I'm not looking for someone elses work, I'm developing an application from scratch.

Michael
2006/8/24, Andrew Kirch [EMAIL PROTECTED]:




Umm… Flash operator panel?

Andrew





From: 
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] 
On Behalf Of MirSent: Thursday, August 24, 2006 2:18 PMTo: 
asterisk-users@lists.digium.com; asterisk-dev@lists.digium.com
Subject: [asterisk-dev] Phone status



Hi



I'm working on a project, where I need the status of every telephone on the system. (Idle,ringing,busy)

If a phone is busy, I also need to know the callerid of the other end.



I have made a deamon, which query Asterisk every second for active calls, this works by issuing a Status to the manager-interface, and processing the return data and then put the result into a MySQLtable. 




The clients will query the MySQL table every second for the state of their phone, if there are no records with their numbers in it, they are considered idle.




This works fine for calls from one SIP-phone to the other, this is for instance what it look like when extension 310 is connected to extension 311:




Event: StatusPrivilege: CallChannel: SIP/310-08697fb8CallerID: 310CallerIDName: unknownAccount: State: UpLink: SIP/311-0868fd98
Uniqueid: 1156442804.74
Event: StatusPrivilege: CallChannel: SIP/311-0868fd98CallerID: 311CallerIDName: SnomAccount: State: UpContext: macro-vm
Extension: sPriority: 5Seconds: 13Link: SIP/310-08697fb8 Uniqueid: 1156442804.73
That is pretty easy to decode.
However when an external call is made to a SIP-phone, the result is different, this is a call from another Asterisk via an IAX trunk:

Event: StatusPrivilege: CallChannel: SIP/311-08695698CallerID: 35254390CallerIDName: unknownAccount: State: UpLink: IAX2/MR-1
Uniqueid: 1156442974.76
Event: StatusPrivilege: CallChannel: IAX2/MR-1CallerID: 35436121CallerIDName: unknownAccount: State: UpContext: macro-vm
Extension: sPriority: 5Seconds: 9Link: SIP/311-08695698 Uniqueid: 1156442974.75
The actual callerid of the caller is 3536121, 35254390 is the called number.
How do I get the information, that 35436121 is connected to 311?
Am I doing it in a stupid way, I'm aware that the Manager can give me realtime events, but I'm under the impression, that it is not very stable in a high traffic environment?

Any help or good ideas would be appriceated.
Michael





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Re: [asterisk-users] Phone status

[Earl Terwilliger]

Michael,

you might take a look at this, as it does most of that + more:
 
http://micpc.com/eventmonitor/

earl

On Thursday 24 August 2006 14:17, Mir wrote:
 Hi

 I'm working on a project, where I need the status of every telephone on the
 system. (Idle,ringing,busy)
 If a phone is busy, I also need to know the callerid of the other end.

 I have made a deamon, which query Asterisk every second for active calls,
 this works by issuing a Status to the manager-interface, and processing
 the return data and then put the result into a MySQL table.

 The clients will query the MySQL table every second for the state of their
 phone, if there are no records with their numbers in it, they are
 considered idle.

 This works fine for calls from one SIP-phone to the other, this is for
 instance what it look like when extension 310 is connected to extension
 311:


 Event: Status
 Privilege: Call
 Channel: SIP/310-08697fb8
 CallerID: 310
 CallerIDName: unknown
 Account:
 State: Up
 Link: SIP/311-0868fd98
 Uniqueid: 1156442804.74


 Event: Status
 Privilege: Call
 Channel: SIP/311-0868fd98
 CallerID: 311
 CallerIDName: Snom
 Account:
 State: Up
 Context: macro-vm
 Extension: s
 Priority: 5
 Seconds: 13
 Link: SIP/310-08697fb8
 Uniqueid: 1156442804.73

 That is pretty easy to decode.

 However when an external call is made to a SIP-phone, the result is
 different, this is a call from another Asterisk via an IAX trunk:

 Event: Status
 Privilege: Call
 Channel: SIP/311-08695698
 CallerID: 35254390
 CallerIDName: unknown
 Account:
 State: Up
 Link: IAX2/MR-1
 Uniqueid: 1156442974.76


 Event: Status
 Privilege: Call
 Channel: IAX2/MR-1
 CallerID: 35436121
 CallerIDName: unknown
 Account:
 State: Up
 Context: macro-vm
 Extension: s
 Priority: 5
 Seconds: 9
 Link: SIP/311-08695698
 Uniqueid: 1156442974.75

 The actual callerid of the caller is 3536121, 35254390 is the called
 number.

 How do I get the information, that 35436121 is connected to 311?

 Am I doing it in a stupid way, I'm aware that the Manager can give me
 realtime events, but I'm under the impression, that it is not very stable
 in a high traffic environment?

 Any help or good ideas would be appriceated.

 Michael
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[asterisk-users] Re: Working Sipura 3000 or Linksys 3102 configuration?

[Vincent Delporte]

At 15:12 23/08/2006 -0700, Ron Wellsted wrote:

Here are mine (with UK regional settings/A-law).


Thanks a bunch :-) After more search, it turns out that if you don't need 
the router feature of the 3102 (I already have a router), the unit must be 
connected to the LAN through its... WAN plug. It would have been nice that 
the pathetic 6-page brochur - which is the only known documentation to date 
for the 3102 - mention that :-/


Next steps:
- setting up an IP phone accross the Internet, and handle the firewall issue
- see if the Linksys can connect out to a remote IP phone directly, with no 
PBX at all.


Cheers
VD.


--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.405 / Virus Database: 268.11.5/426 - Release Date: 23/08/2006


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Re: [asterisk-users] Re: Zaptel install - Fedora Core 5

[Bruno Wolff III]

On Tue, Aug 22, 2006 at 11:56:26 +0200,
  Tomislav Parčina [EMAIL PROTECTED] wrote:
 
 The thing is that I can't rely on yum update for asterisk installation. I 
 would like something that will work like this: I install FC5 from CD/DVD, 
 install RPM's that I need from my ftp server or from CD, install zaptel, 
 libpri, asterisk...
 
 So, I need to download rpm's that will allow me to install 
 zaptel/libpri/asterisk without using yum update (I need to make all 
 installations the same).

There are FC5 compatible RPMS at:
http://www.xs4all.nl/~pjl/downloads/asterisk/

I have been using SRPMS there to make new zaptel RPMS when FC5 gets kernel
updates. I haven't tried the 1.2.11 release that was just posted, but
earlier versions seemed to work fine.
Note those aren't vanilla releases. There are a few patches applied.
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RE: [asterisk-users] RE: [asterisk-dev] Phone status

[Dean Collins]

I guess what Andrew was saying is what are
you trying to do specifically that Flash Operator Panel doesnt already give
you







Cheers,

Dean













From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mir
Sent: Thursday, 24 August 2006
2:48 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-users] RE:
[asterisk-dev] Phone status







What do you mean?











I'm not looking for someone elses work, I'm developing an application
from scratch.











Michael







2006/8/24, Andrew Kirch [EMAIL PROTECTED]:








Umm Flash operator panel?



Andrew











From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]
On Behalf Of Mir
Sent: Thursday, August 24, 2006
2:18 PM
To: asterisk-users@lists.digium.com;
asterisk-dev@lists.digium.com
Subject: [asterisk-dev] Phone
status











Hi











I'm
working on a project, where I need the status of every telephone on the system.
(Idle,ringing,busy)





If
a phone is busy, I also need to know the callerid of the other end.











I
have made a deamon, which query Asterisk every second for active calls, this
works by issuing a Status to the manager-interface, and processing
the return data and then put the result into a MySQLtable. 











The
clients will query the MySQL table every second for the state of their phone,
if there are no records with their numbers in it, they are considered idle. 











This
works fine for calls from one SIP-phone to the other, this is for instance what
it look like when extension 310 is connected to extension 311: 











Event:
Status
Privilege: Call
Channel: SIP/310-08697fb8
CallerID: 310
CallerIDName: unknown
Account: 
State: Up
Link: SIP/311-0868fd98 
Uniqueid: 1156442804.74


Event: Status
Privilege: Call
Channel: SIP/311-0868fd98
CallerID: 311
CallerIDName: Snom
Account: 
State: Up
Context: macro-vm 
Extension: s
Priority: 5
Seconds: 13
Link: SIP/310-08697fb8 
Uniqueid: 1156442804.73

That
is pretty easy to decode.

However
when an external call is made to a SIP-phone, the result is different, this is
a call from another Asterisk via an IAX trunk:

Event:
Status
Privilege: Call
Channel: SIP/311-08695698
CallerID: 35254390
CallerIDName: unknown
Account: 
State: Up
Link: IAX2/MR-1 
Uniqueid: 1156442974.76


Event: Status
Privilege: Call
Channel: IAX2/MR-1
CallerID: 35436121
CallerIDName: unknown
Account: 
State: Up
Context: macro-vm 
Extension: s
Priority: 5
Seconds: 9
Link: SIP/311-08695698 
Uniqueid: 1156442974.75

The
actual callerid of the caller is 3536121, 35254390 is the called number.

How
do I get the information, that 35436121 is connected to 311?

Am
I doing it in a stupid way, I'm aware that the Manager can give me realtime
events, but I'm under the impression, that it is not very stable in a high
traffic environment? 

Any
help or good ideas would be appriceated.

Michael


























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[asterisk-users] Re: Nokia E60/61/70 and SIP

[Benny Amorsen]

 H == Haspers  [EMAIL PROTECTED] writes:

H I've got them all. It registers correctly with Asterisk, and get
H incoming calls, but it complaints about outgoing calls (Connection
H Error). SIP Debug is giving me: SIP/2.0 407 Proxy Authentication
H Required
 
H But those settings are the same (Proxy Server/Registrar Server). So
H what could be the problem?

I never put anything in the Proxy settings, only in the Registrar
settings. Maybe that makes a difference?


/Benny


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Re: [asterisk-users] voicemailmain

[Doug Lytle]

Aaron Daniel wrote:

Not sure about that Doug.  It should read:

exten = a,1,VoicemailMan([EMAIL PROTECTED])
  


You are correct.

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] Static in Monitor recordings

[Adam Kavan]

I turned off hyper threading and it did
not help, I also tried telling it to record as gsm files but it still makes an
awful noise. Anyone have any other ideas?











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Don
Sent: Friday, August 18, 2006
12:04 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-users]
Static in Monitor recordings







Sounds like a poweredge server and the 2850 we
have...hyperthreading has caused all kinds of crazy stuff...and turning it off
just in grub.conf doesn't solve it until it is turned off in the bios.







- Original Message - 





From: Adam Kavan 





To: asterisk-users@lists.digium.com 





Sent: Friday, August 18,
2006 12:57 PM





Subject: [asterisk-users]
Static in Monitor recordings









I am running Asterisk 1.2.9.1 in a call center with 26
agents placing outbound calls using SIP soft phones going out a Diginum 4 port
T1 card (All 4 spans have PRI t1s).



All of the calls run through 



[macro-record-call]

exten =
s,1,Monitor(wav,${ARG1}-${CALLERIDNUM}-${DATETIME:0:11}-${DATETIME:12:2}-${DATETIME:15:2},mb)



and this works fairly well, however several times a day I
get recordings with static in them. It sounds like a corrupted wav file,
with noise in it all over the place. It is obvious from the conversations
occurring in the recording that the other people involved in the call do not
hear any of the static, so I assume that it is coming from the recordings.



Also, our QA people are using ChanSpy to listen into these
calls and tell me that they can hear the static as well, but I have not been
able to verify there claims.



Last night I tried using MixMonitor instead of just Monitor
and QA claims that there were no problems with the sound, but my Asterisk
server kept crashing and I assume that MixMonitor was at fault (the crashes
went away as soon as I changed back to regular Monitor).



Does anyone know why this might be happening?



On the whole I am having no stability problems other than
last night with MixMonitor. I eventually want to scale my deployment up
toa little over 200 agents and I want to make sure I get this figured out while
I still am using my fairly small testing population.



My server is a Dell with 2 X Intel(R) Xeon(TM) CPU 3.20GHz
with hyperthreading turned on and 4 GM of RAM and my storage is a SCSI hardware
raid card, pointing to a mirror.



Any help you can give me would be greatly appreciated.



--- Adam Kavan

--- [EMAIL PROTECTED]







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No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.405 / Virus Database: 268.11.3/423 - Release Date: 8/18/2006








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[asterisk-users] voicemailmain

[existx]

Howdy,

I have a Debian box using Debian's Asterisk package. People can leave
voicemail for the extensions that are setup in the configuration, and
asterisk e-mail's the user a .wav file (voicemail.conf). This works
perfect.

However, I want to have VoicemailMain sit on an extension so people
can call in, change their greeting, listen too voicemail, etc.

extensions.conf:

exten = 2999,1,Answer
exten = 2999,2,Wait,2
exten = 2999,3,Voicemailmain()

My understand is, that this should allow any user to call up. Enter in
their mailbox number (currently the same as their extension) and
password. However, I cannot dial this extension after reloading
asterisk.

I'm thinking I should add something in another configuration file, or
perhaps my syntax is wrong. Any help would be much apperciated!

Thanks in advance.

Regards,
Jason
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Re: [asterisk-users] voicemailmain

[Hadley Rich]

On Friday 25 August 2006 08:39, existx wrote:
 The error from the CLI is:

 Aug 24 16:13:49 NOTICE[23174]: chan_iax2.c:7241 socket_read: Rejected
 connect attempt from 192.168.0.23, request '[EMAIL PROTECTED]' does not
 exist

It looks like you have created 2699 in a different context than your phones. 
You will need to include = the-context to be able to dial the extension.

-- 
http://nicegear.co.nz
New Zealand's VoIP supplier
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Re: [asterisk-users] Asterisk t38passthrough

[William Piper]

Perhaps a stupid suggestion... but did you make sure that the ATA had the T38 selected in the GUI?

bp
On 8/24/06, Ricardo Carvalho [EMAIL PROTECTED] wrote:
Hi,I've installed Asterisk t38passthrough branch and I'm using oneGrandstream ATA to connect Asterisk to a Fax machine. Every time I send
a fax, it gets sent using codec G711, and never T.38. I added thefollowing parameters in the [general] section as well as in deviceconfigurations:t38pt_udptl = yest38pt_rtp = yest38pt_tcp = yes
I think that's the only thing that is needed to do to enable T.38 passthrough...Why does Asterisk keeps sending in G711? Any help?Regards,Ricardo.___
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Re: [asterisk-users] voicemailmain

[existx]

Cristian,

The only other line in extensions.conf that references VoicemailMain is this:

exten = a,1,VoicemailMain(${ARG1})

The error from the CLI is:

Aug 24 16:13:49 NOTICE[23174]: chan_iax2.c:7241 socket_read: Rejected
connect attempt from 192.168.0.23, request '[EMAIL PROTECTED]' does not
exist

Regards,
Jason



On 8/24/06, kritikus Araklidas [EMAIL PROTECTED] wrote:

Hi:

First it at all check if you have a different extension for voicemailmain.?

Then use VoiceMailMain syntax.

And send me the CLI log when you try to connect to VoiceMailMain.

regards.

Cristian.


From: existx [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] voicemailmain
Date: Thu, 24 Aug 2006 16:08:01 -0400

Howdy,

I have a Debian box using Debian's Asterisk package. People can leave
voicemail for the extensions that are setup in the configuration, and
asterisk e-mail's the user a .wav file (voicemail.conf). This works
perfect.

However, I want to have VoicemailMain sit on an extension so people
can call in, change their greeting, listen too voicemail, etc.

extensions.conf:

exten = 2999,1,Answer
exten = 2999,2,Wait,2
exten = 2999,3,Voicemailmain()

My understand is, that this should allow any user to call up. Enter in
their mailbox number (currently the same as their extension) and
password. However, I cannot dial this extension after reloading
asterisk.

I'm thinking I should add something in another configuration file, or
perhaps my syntax is wrong. Any help would be much apperciated!

Thanks in advance.

Regards,
Jason
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Re: [asterisk-users] voicemailmain

[Doug Lytle]

existx wrote:

Cristian,

The only other line in extensions.conf that references VoicemailMain 
is this:


exten = a,1,VoicemailMain(${ARG1})


This should read:

exten = a,1,VoicemailMain([EMAIL PROTECTED])


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] voicemailmain

[kritikus Araklidas]

Ok you have two optionsthe iax extension is created under default 
context???


The VoceMilMain could be configured with the options of wich context use 
like this:


extensions.conf:

exten = 2999,1,Answer
exten = 2999,2,Wait,2
exten = 2999,3,Voicemailmain(@test)

Where test is the context where the iax client belong.

Let me know.

Chers.

Cris.





From: existx [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com

Subject: Re: [asterisk-users] voicemailmain
Date: Thu, 24 Aug 2006 16:39:35 -0400

Cristian,

The only other line in extensions.conf that references VoicemailMain is 
this:


exten = a,1,VoicemailMain(${ARG1})

The error from the CLI is:

Aug 24 16:13:49 NOTICE[23174]: chan_iax2.c:7241 socket_read: Rejected
connect attempt from 192.168.0.23, request '[EMAIL PROTECTED]' does not
exist

Regards,
Jason



On 8/24/06, kritikus Araklidas [EMAIL PROTECTED] wrote:

Hi:

First it at all check if you have a different extension for 
voicemailmain.?


Then use VoiceMailMain syntax.

And send me the CLI log when you try to connect to VoiceMailMain.

regards.

Cristian.


From: existx [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] voicemailmain
Date: Thu, 24 Aug 2006 16:08:01 -0400

Howdy,

I have a Debian box using Debian's Asterisk package. People can leave
voicemail for the extensions that are setup in the configuration, and
asterisk e-mail's the user a .wav file (voicemail.conf). This works
perfect.

However, I want to have VoicemailMain sit on an extension so people
can call in, change their greeting, listen too voicemail, etc.

extensions.conf:

exten = 2999,1,Answer
exten = 2999,2,Wait,2
exten = 2999,3,Voicemailmain()

My understand is, that this should allow any user to call up. Enter in
their mailbox number (currently the same as their extension) and
password. However, I cannot dial this extension after reloading
asterisk.

I'm thinking I should add something in another configuration file, or
perhaps my syntax is wrong. Any help would be much apperciated!

Thanks in advance.

Regards,
Jason
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RE: [asterisk-users] voicemailmain

[kritikus Araklidas]

Hi:

First it at all check if you have a different extension for voicemailmain.?

Then use VoiceMailMain syntax.

And send me the CLI log when you try to connect to VoiceMailMain.

regards.

Cristian.



From: existx [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com

To: asterisk-users@lists.digium.com
Subject: [asterisk-users] voicemailmain
Date: Thu, 24 Aug 2006 16:08:01 -0400

Howdy,

I have a Debian box using Debian's Asterisk package. People can leave
voicemail for the extensions that are setup in the configuration, and
asterisk e-mail's the user a .wav file (voicemail.conf). This works
perfect.

However, I want to have VoicemailMain sit on an extension so people
can call in, change their greeting, listen too voicemail, etc.

extensions.conf:

exten = 2999,1,Answer
exten = 2999,2,Wait,2
exten = 2999,3,Voicemailmain()

My understand is, that this should allow any user to call up. Enter in
their mailbox number (currently the same as their extension) and
password. However, I cannot dial this extension after reloading
asterisk.

I'm thinking I should add something in another configuration file, or
perhaps my syntax is wrong. Any help would be much apperciated!

Thanks in advance.

Regards,
Jason
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Re: [asterisk-users] voicemailmain

[Aaron Daniel]

Not sure about that Doug.  It should read:

exten = a,1,VoicemailMan([EMAIL PROTECTED])

If you put it in the brackets, it becomes part of the variable name
instead of part of the argument.

On Thu, 2006-08-24 at 16:57 -0400, Doug Lytle wrote:
 existx wrote:
  Cristian,
 
  The only other line in extensions.conf that references VoicemailMain 
  is this:
 
  exten = a,1,VoicemailMain(${ARG1})
 
 This should read:
 
 exten = a,1,VoicemailMain([EMAIL PROTECTED])
 
 
 Doug
 
-- 
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198

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Re: [asterisk-users] voicemailmain

[existx]

Howdy guys,

Thanks for your help, it works fine without editing the default line of:

exten = a,1,VoicemailMain(${ARG1})

The issue was that I had specified VoicemailMain by the default line,
which was way above the rest of my extensions (out of context).

Hopefully this will help someone in the future.

Regards,
Jason



On 8/24/06, Aaron Daniel [EMAIL PROTECTED] wrote:

Not sure about that Doug.  It should read:

exten = a,1,VoicemailMan([EMAIL PROTECTED])

If you put it in the brackets, it becomes part of the variable name
instead of part of the argument.

On Thu, 2006-08-24 at 16:57 -0400, Doug Lytle wrote:
 existx wrote:
  Cristian,
 
  The only other line in extensions.conf that references VoicemailMain
  is this:
 
  exten = a,1,VoicemailMain(${ARG1})

 This should read:

 exten = a,1,VoicemailMain([EMAIL PROTECTED])


 Doug

--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198

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[asterisk-users] Polycom microbrowser issue Error HTTP 406 with IIS

[Phil Menico]

Title: Message




I have 
no where else to turn to so if anyone has an answer please send 
it.

I am running sip version 1.6.on a Polycom 601on 
Asterisk and am unable to get the microbroser to work. The phone returns 
a 406 error for both idle and services. 
I can see the file being requested and the subsequent 406 error in the IIS 
log files. Any ideas on what permissions are needed in IIS or how to format 
the webpage file?
I 
tried both these 2 files with no luck

XHTML 
file 1:

html head 
/head body Hello phil 
post /body/html


XHTML 
file 2:

?xml version="1.0" 
encoding="UTF-8"?html xmlns="http://www.w3.org/1999/xhtml" 
xml:lang="en" lang="en" head 
titleVirtual Library/title /head 
body PHello phil/P 
/body/html

Log 
info from IIS:

2006-08-24 20:39:18 10.0.3.175 - 
W3SVC1 PHIL3 10.0.1.210 81 GET /Polycom/ - 302 0 295 202 0 HTTP/1.1 
10.0.1.210:81 
Polycom-Microbrowser/1.0+(SIP/1.6.3.0067;+SoundPoint+IP+PolycomSoundPointIP-SPIP_601)+libcurl/7.12.1 
- -2006-08-24 20:39:18 10.0.3.175 - W3SVC1 PHIL3 10.0.1.210 81 GET 
/Polycom/post.htm - 406 0 4085 242 10 HTTP/1.1 10.0.1.210:81 
Polycom-Microbrowser/1.0+(SIP/1.6.3.0067;+SoundPoint+IP+PolycomSoundPointIP-SPIP_601)+libcurl/7.12.1 
- http://10.0.1.210:81/Polycom

Thank you.
Phil 

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Re: [asterisk-users] Modems dialing over sangoma a104d

[Wolfgang Zweimueller]

Rich Adamson [EMAIL PROTECTED] writes:

 Sean Cook wrote:
 I have a sangoma 104d that is our main pbx now( legacy system died ).  I
 have replaced every phone in the building and things are going very well.
 We have fax working well and calls are routing properly...  All is well...

 Except for our support modems... we have support people that dial out with
 modems across our PRI's.  These modems are attached to an Adtran 750 with 24
 FXS's.  I have disabled echo cancelation on the T1 that is connected to the
 Adtran but negotiation is still really rough.  I am bridging across the same
 card and it isn't doing very well... has anyone done this with reasonably
 successful results?  I am not looking for 56K I am looking for around 9600
 to 14.4..

 Can we assume that you've got the correct timing parameters set on the
 104d?  (eg, are you sync'ing your 104d from the telco?)

 If not, get that corrected first as it makes a major difference with
 modem calls.

That's the point! We had the same issues: modem calls dropping after a
few minutes. Get at least wanpipe-beta7-2.3.4.tgz and set the
reference clock to the telco line and set MASTER to this line.


cu,
Wolfgang
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[asterisk-users] Snom phones locking up

[garth]

Hi All

I have had a problem with a few Snom 320's on several sites locking up
after a few days.  I am running application ver 6.2.2 with the latest
jffs2 ver and tried the latest 5.x ver with similar results.  Is this also
experienced with other Snom users?

I know some posts say it could be the network switches etc, but Cisco?  I
fail to see how a switch could bring down a device.

Kind Regards
Garth


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[asterisk-users] Idiot questions

[joea, j4computers]

As a complete newcomer to Asterisk, Digium and PBX, I have several questions.

But I'll start with this.

To setup a simple system with only a couple of POTS lines, I gather I will need 
a TDM400 board with FXO and/or FXS modules.

So, a TDM400 card will support up to two analog (POTS) lines?

joea
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Re: [asterisk-users] Idiot questions

[Ron McCarthy]

FXO is coming from the PSTN, FXS is what devices connect to (like a analog phone).If you are using VOIP phone then you dont need the FXS modules, just FXO.On 8/24/06, 
joea, j4computers [EMAIL PROTECTED] wrote:
As a complete newcomer to Asterisk, Digium and PBX, I have several questions.But I'll start with this.To setup a simple system with only a couple of POTS lines, I gather I will need a TDM400 board with FXO and/or FXS modules.
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[asterisk-users] Attempt to setup paging and intercom

[Larry Alkoff]

This is my first attempt to setup intercom and paging for some Grandview 
sip phones per instructions from Grandview.


I put the lines below in extensions.conf and did the CLI reload command.

When I issue
**1 or **2 from a phone I get a 404 error.
Shouldn't that be ringing the 3 phones on my list?

The instructions are a little vague (to a newbie like me) and may well 
be wrong.


Here is what I put in extensions.conf:

--  Stop reading here if not interested   

; from: FAQ_Asterisk_Paging_for_GXP-2000.pdf

; Paging and Intercom:
; 
; Grandstream Phone Configuration:
;   Allow Auto Answer by Call-Info: Yes
;   Turn off speaker on remote disconnect:  Yes

; Note: Above configuration will allow GXP-2000 to auto answer a call
; when the call contains:
;  SIP header Call-Info: answer-after=0
; And when the call hung up by the remote party,
; the phone will automatically on hook without alerting user with
; disconnect busy tones.

; Asterisk Configuration:
; ===
; Then you can set up Asterisk with following functions:

; 1) One to One Intercom
; ==

; You will first define a Macro and then use it in the one to one 
intercom context:

[macro-pageext]
exten = s,1,ChanIsAvail(${ARG1}|js) ; j is for dump and s is for ANY call
exten = s,2,SIPAddHeader(Call-Info: answer-after=0)
exten = s,3,Dial(${ARG1})
exten = s,4,NoOp() ; Add others here
exten = s,5, Hangup
exten = s,102,Hangup

[INTERCOM_GROUP]
exten = _*1XX,1,Macro(pageext,SIP/${EXTEN:1}) ;Page each extension
exten = _*1XX,2,Hangup
; Note: Above configuration will allow user intercom with any extension
; (using 1XX) by dialing *1XX.

; 2) One to Many Paging
; =

[One_Way_Page_GROUP]
exten = _**1,1,SIPAddHeader(Call-Info: answer-after=0)
exten = _**1,2,Page(${One_Way_Paging_List}|)
exten = _**1,3, Hangup
; Note: Above configuration will allow user to one way page(broadcast)
; to all
; the extensions defined in variable One_Way_Paging_list
; which can be define as following:

One_Way_Paging_List = SIP/120SIP/122/SIP/100

; 3) One to Many Intercom
; ===

[Two_Way_Intercom_GROUP]
exten = _**2,1,SIPAddHeader(Call-Info: answer-after=0)
exten = _**2,2,Page(${Two_Way_Intercom_List}|d)
exten = _**2,3, Hangup
; Note: Above configuration will allow user to do two way intercom to 
all the

; extensions defined in variable Two_Way_Intercom_List which can be
; define as following:

Two_Way_Intercom_List = SIP/120SIP/122/SIP/100

--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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Re: [asterisk-users] Idiot questions

[Thomas Artner]

joea, j4computers wrote:
 As a complete newcomer to Asterisk, Digium and PBX, I have several questions.
 
 But I'll start with this.
 
 To setup a simple system with only a couple of POTS lines, I gather I will 
 need a TDM400 board with FXO and/or FXS modules.
 
 So, a TDM400 card will support up to two analog (POTS) lines?

a tdm400 card has 4 slots. each of these slots can be assembled with a
FXS or FXO module.

So you can handle 4 FXO lines, or 4 FXS, or 2FXO and 2 FXS 

but its recommended to use only one tdm400 card per computer.


 
 joea
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Re: [asterisk-users] Snom phones locking up

[Peter J Dean]

I have had a problem with a few Snom 320's on several sites locking up
after a few days.  I am running application ver 6.2.2 with the latest
jffs2 ver and tried the latest 5.x ver with similar results.  Is  
this also

experienced with other Snom users?



not sure if this will help you identify the lock-ups (regardless of  
v6.2.2 or v6.2.3 same results) I have found that issuing a,


sip notify reboot-snom extension number causes the phone to go into  
a weird state - which could almost be defined as a lock-up state, and  
the only way to reset it is to remove power.


after modifying the sip_notify.conf file,

from,
[reboot-snom]
Event=reboot
Content-Length=0

to (don't forget to do a sip reload),
[reboot-snom]
Event=check-sync;reboot=false
Content-Length=0

then re-issued a sip notify reboot-snom extension number the phone  
reboots fine.




I know some posts say it could be the network switches etc, but  
Cisco?  I

fail to see how a switch could bring down a device.



We use allied telesyn switches.



smime.p7s
Description: S/MIME cryptographic signature
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Re: [asterisk-users] Idiot questions

[Mojo with Horan Company, LLC]

a TDM400 card will hold up to four modules, which can be FXO or FXS. 
depending on the modules purchased you can connect up to four phones 
(using FXS modules) or four incoming phone lines (using FXO modules), or 
any combination thereof.


joea, j4computers wrote:

As a complete newcomer to Asterisk, Digium and PBX, I have several questions.

But I'll start with this.

To setup a simple system with only a couple of POTS lines, I gather I will need 
a TDM400 board with FXO and/or FXS modules.

So, a TDM400 card will support up to two analog (POTS) lines?

joea
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!DSPAM:500,44ee20ea208361132923654!



--
Mojo [EMAIL PROTECTED]
Office Manager, Horan  Company, LLC
(907) 747- x112
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Re: [asterisk-users] Idiot questions

[Adam Collard]

You will need a TDM400 with an FXO module for each line you want. A TDM400 
supports up to four lines or analog stations. For two lines, you should get a 
TDM04B.
-Original message-
From: joea, j4computers [EMAIL PROTECTED]
Date: Thu, 24 Aug 2006 14:58:21 -0700
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Idiot questions

 As a complete newcomer to Asterisk, Digium and PBX, I have several questions.
 
 But I'll start with this.
 
 To setup a simple system with only a couple of POTS lines, I gather I will 
 need a TDM400 board with FXO and/or FXS modules.
 
 So, a TDM400 card will support up to two analog (POTS) lines?
 
 joea
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Adam Collard
President
Digital Telecom of Michigan, Inc.
[EMAIL PROTECTED]
(517) 233-1072 Direct Office
(800) 420-3803 x4101 Office
(517) 766-5902 Fax

This email may be confidential. Any distribution, use or copying of this email 
or the information it contains by other than an intended recipient is 
unauthorized. If you received this email in error, please advise me (by return 
email or otherwise) immediately.

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RE: [asterisk-users] Asterisk t38passthrough

[Edgar Barbosa]

Also, make sure you have
a T.38 enabled device at the other end 



Edgar













From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of William Piper
Sent: quinta-feira, 24 de Agosto
de 2006 21:09
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-users]
Asterisk t38passthrough







Perhaps a stupid suggestion... but did you make sure that the ATA had
the T38 selected in the GUI?











bp







On 8/24/06, Ricardo
Carvalho [EMAIL PROTECTED]
wrote: 

Hi,

I've installed Asterisk t38passthrough branch and I'm using one
Grandstream ATA to connect Asterisk to a Fax machine. Every time I send 
a fax, it gets sent using codec G711, and never T.38. I added the
following parameters in the [general] section as well as in device
configurations:

t38pt_udptl = yes
t38pt_rtp = yes
t38pt_tcp = yes 


I think that's the only thing that is needed to do to enable T.38 pass
through...
Why does Asterisk keeps sending in G711? Any help?

Regards,

Ricardo.
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RE: [asterisk-users] Polycom microbrowser issue Error HTTP 406 with IIS

[Douglas Garstang]

Title: Message



We had 
a similar problem. Eventuallywe gave up and just used apache. We found 
that _exactly_ the same content would not work with IIS, but WOULD work with 
Apache.

  -Original Message-From: Phil Menico 
  [mailto:[EMAIL PROTECTED]Sent: Thursday, August 24, 2006 3:06 
  PMTo: asterisk-users@lists.digium.comSubject: 
  [asterisk-users] Polycom microbrowser issue Error HTTP 406 with 
  IIS
  
  I 
  have no where else to turn to so if anyone has an answer please send 
  it.
  
  I am running sip version 1.6.on a Polycom 601on 
  Asterisk and am unable to get the microbroser to work. The phone 
  returns a 406 error for both idle and 
  services. I can see the file being requested and the subsequent 406 
  error in the IIS log files. Any ideas on what permissions are needed in 
  IIS or how to format the webpage file?
  I 
  tried both these 2 files with no luck
  
  XHTML file 1:
  
  html head 
  /head body Hello phil 
  post /body/html
  
  
  XHTML file 2:
  
  ?xml version="1.0" 
  encoding="UTF-8"?html xmlns="http://www.w3.org/1999/xhtml" 
  xml:lang="en" lang="en" head 
  titleVirtual Library/title /head 
  body PHello phil/P 
  /body/html
  
  Log 
  info from IIS:
  
  2006-08-24 20:39:18 10.0.3.175 - 
  W3SVC1 PHIL3 10.0.1.210 81 GET /Polycom/ - 302 0 295 202 0 HTTP/1.1 
  10.0.1.210:81 
  Polycom-Microbrowser/1.0+(SIP/1.6.3.0067;+SoundPoint+IP+PolycomSoundPointIP-SPIP_601)+libcurl/7.12.1 
  - -2006-08-24 20:39:18 10.0.3.175 - W3SVC1 PHIL3 10.0.1.210 81 GET 
  /Polycom/post.htm - 406 0 4085 242 10 HTTP/1.1 10.0.1.210:81 
  Polycom-Microbrowser/1.0+(SIP/1.6.3.0067;+SoundPoint+IP+PolycomSoundPointIP-SPIP_601)+libcurl/7.12.1 
  - http://10.0.1.210:81/Polycom
  
  Thank you.
  Phil 

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RE: [asterisk-users] Call Parking Ring Back (Snoms)

[David Gagnon]

Look over there : http://bugs.digium.com/view.php?id=6953

David

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de J. Oquendo
Envoyé : 24 août 2006 13:54
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : [asterisk-users] Call Parking Ring Back (Snoms)

Quick question maybe someone can point me in the right direction...

Caller -- Receptionist -- ParksCall
Receptionist makes announcement for individual to pick up parked call. 
No one picks up so it rings back to receptionist within a minute and a 
half. Is there any way to change the ringer for a parked call coming 
back since their call wasn't answered?

-- 

J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743
sil . infiltrated @ net http://www.infiltrated.net 

The happiness of society is the end of government.
John Adams

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Re: [asterisk-users] SpanDSP Error

[Christian Jensen]

Well I got it resolved. Thanks steve for telling me. I looked all  
around the filesystem and found some files I did not delete from  
spandsp0.0.3. I saw that solution before and tried it and it didn't  
seem to work and I wondered why but now i know. Thank you.

Chris
On Aug 24, 2006, at 10:54 AM, Steve Underwood wrote:


Christian Jensen wrote:

I have not found any solution to the problem I am talking about  
in  the archives Steve. I have reinstalled and downgraded from  
pre26 to  pre21 of  SpanDSP 0.0.2 and still to no avail. I have  
followed all  the instructions and ways of fixing this problem and  
have found none  to be a solution.

-configured with prefix /usr
-make
-make install
-then moved the patch file and the corresponding .c files into  
the  apps directory of my asterisksource
-Then patched the Makefile and recompiled asterisk. I have done  
it  with about 4 versions so far of which I cannot recall which  
ones.  Pre26 and Pre21 definately

Anyone?


If you look, you will find many occurrences of the answer. Remove  
spandsp 0.0.3 from your system. It is there, and it is causing  
conflicts.


Steve

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[asterisk-users] hint status not updating on inbound

[Damon Estep]

I have the hint priority defined for a few SIP
phones.



When I make a call OUT from one of the phones I see that the
show hints picks up a status change from 0 to 1 for the extension,
but when I call IN to that extension the hint status is still 0.



This is on a server built back in September of 2005. It has
been very reliable (and busy) so I do not want to upgrade it if I can avoid it.



Does anyone know if this was a bug at one time? 



Could there be something in my config that is causing this
behavior?










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Re: [asterisk-users] No outbound with A2Billing

[Leo Ann Boon]

Luciano Moreira wrote:

List members,

When I dial to a PSTN number, the A2Billing script does all the tasks,
until it shutdown without make the dailout by sip trunk set.
Lasts outputs fro the a2billing.php debug are:
  a2billing.php|2: RESFINDRATE:: 0
  a2billing.php|2: UPDATE cc_card SET inuse=inuse-1 WHERE username='5033845534'

  

Luciano,

First, this is a specific a2billing question not an Asterisk question. 
You should post it a2billing mailing list.


As for your problem: you can't call because you haven't defined a rate 
for that destination prefix in A2billing. That's why you get 
RESFINDRATE:: 0.


Leo.

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Re: [asterisk-users] Idiot questions

[Nilesh Londhe]

I would suggest buying a very lowprice FXO to begin with which would probably be x100p PCI card at ebay for about $10 +shipping.
On 8/24/06, Adam Collard [EMAIL PROTECTED] wrote:
You will need a TDM400 with an FXO module for each line you want. A TDM400 supports up to four lines or analog stations. For two lines, you should get a TDM04B.
-Original message-From: joea, j4computers [EMAIL PROTECTED]Date: Thu, 24 Aug 2006 14:58:21 -0700To: 
asterisk-users@lists.digium.comSubject: [asterisk-users] Idiot questions As a complete newcomer to Asterisk, Digium and PBX, I have several questions. But I'll start with this.
 To setup a simple system with only a couple of POTS lines, I gather I will need a TDM400 board with FXO and/or FXS modules. So, a TDM400 card will support up to two analog (POTS) lines? joea
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http://lists.digium.com/mailman/listinfo/asterisk-usersAdam CollardPresidentDigital Telecom of Michigan, Inc.
[EMAIL PROTECTED](517) 233-1072 Direct Office(800) 420-3803 x4101 Office(517) 766-5902 FaxThis email may be confidential. Any distribution, use or copying of this email or the information it contains by other than an intended recipient is unauthorized. If you received this email in error, please advise me (by return email or otherwise) immediately.
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Re: [asterisk-users] RE: [asterisk-dev] Phone status

[C F]

So how about inventing a car? The auto industry is much more profitable.

The point; there is no point in reinventing the wheel, why are you
writing this from scratch?

On 8/24/06, Mir [EMAIL PROTECTED] wrote:


What do you mean?

I'm not looking for someone elses work, I'm developing an application from
scratch.

Michael


2006/8/24, Andrew Kirch [EMAIL PROTECTED]:






Umm… Flash operator panel?



Andrew



 


From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Mir
Sent: Thursday, August 24, 2006 2:18 PM
To: asterisk-users@lists.digium.com; asterisk-dev@lists.digium.com
 Subject: [asterisk-dev] Phone status





Hi





I'm working on a project, where I need the status of every telephone on the
system. (Idle,ringing,busy)


If a phone is busy, I also need to know the callerid of the other end.





I have made a deamon, which query Asterisk every second for active calls,
this works by issuing a Status to the manager-interface, and processing
the return data and then put the result into a MySQL table.





The clients will query the MySQL table every second for the state of their
phone, if there are no records with their numbers in it, they are considered
idle.





This works fine for calls from one SIP-phone to the other, this is for
instance what it look like when extension 310 is connected to extension 311:





Event: Status
Privilege: Call
Channel: SIP/310-08697fb8
CallerID: 310
CallerIDName: unknown
Account:
State: Up
Link: SIP/311-0868fd98
Uniqueid: 1156442804.74


Event: Status
Privilege: Call
Channel: SIP/311-0868fd98
CallerID: 311
CallerIDName: Snom
Account:
State: Up
Context: macro-vm
Extension: s
Priority: 5
Seconds: 13
Link: SIP/310-08697fb8
Uniqueid: 1156442804.73

That is pretty easy to decode.

However when an external call is made to a SIP-phone, the result is
different, this is a call from another Asterisk via an IAX trunk:

Event: Status
Privilege: Call
Channel: SIP/311-08695698
CallerID: 35254390
CallerIDName: unknown
Account:
State: Up
Link: IAX2/MR-1
Uniqueid: 1156442974.76


Event: Status
Privilege: Call
Channel: IAX2/MR-1
CallerID: 35436121
CallerIDName: unknown
Account:
State: Up
Context: macro-vm
Extension: s
Priority: 5
Seconds: 9
Link: SIP/311-08695698
Uniqueid: 1156442974.75

The actual callerid of the caller is 3536121, 35254390 is the called number.

How do I get the information, that 35436121 is connected to 311?

Am I doing it in a stupid way, I'm aware that the Manager can give me
realtime events, but I'm under the impression, that it is not very stable in
a high traffic environment?

Any help or good ideas would be appriceated.

Michael









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