[Asterisk-Users] AreskiCC and asterisk

[zhu]

hello, guys:
I have a problem in connecting AreskiCC and asterisk. when i entered the 
numbers, nothing happened and say goodbye. Please tell me how to dial 
the numbers if i am in asia and what is the number format.

Any help would be appreciated!
zhu

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[Asterisk-Users] Local numbers

[jonr]

If I set up an * server will I still be able to use my local Anchorage 
phone number through my * box?


Thanks for any help,

Jon
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[Asterisk-Users] Unable to find a path from g729 to gsm

[Kumara Jayaweera]

Greetings! to all

Now, with some hard time and help from many genurous people's in the list, I
have come to this point with my TDM20B card & my teliax's IAX2 account.



I hope someone may help me with this issue mentioned below. I have already
selected my codec as gms in my iax.conf as well as in teliax's  "my account
page" but still i have the same error when I attempt to make a call.



Second, my last digit is not allowed from teliax. that means I need one more
digit from teliax for dialing through them.



Third, I have somewhat poor support from teliax since I have send them 3,4
emails and so far i got no replies.



Please, help me to go ahead from this point

Thanking all of you,

Kumara.



The error I got



Jun 17 18:47:05 NOTICE[7396]: channel.c:1884 set_format: Unable to find a
path from g729 to gsm
Jun 17 18:47:05 NOTICE[7396]: channel.c:1884 set_format: Unable to find a
path from g729 to gsm
Jun 17 18:47:05 NOTICE[7396]: channel.c:1884 set_format: Unable to find a
path from g729 to gsm
-- IAX2/teliax-1 is ringing
-- IAX2/teliax-1 answered Zap/1-1
Jun 17 18:47:18 WARNING[7396]: channel.c:2308 ast_channel_make_compatible:
No path to translate from Zap/1-1(68) to IAX2/teliax-1(256)
Jun 17 18:47:18 WARNING[7396]: app_dial.c:1324 dial_exec_full: Had to drop
call because I couldn't make Zap/1-1 compatible with IAX2/teliax-1
-- Hungup 'IAX2/teliax-1'
  == Spawn extension (outgoing, 19737228839, 1) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'


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[Asterisk-Users] Queue/How to get the number of incoming calls

[Gary Li]

Hi,all,
 
Now,I am working at make an realtime monitor for the call center based on asterisk.
and ,I had search the archive and wiki.Through the return info from the management API,
I can get the waiting calls ,abandoned calls ,hold time, etc,but I don't know how to get
the number of incoming calls.
The info like following :
 

Asterisk Call Manager/1.0 Response: Success Message: Authentication accepted Response: Success Message: Queue status will follow 
Event: QueueParams Queue: test_queue Max: 18 Calls: 1 Holdtime: 127 Completed: 209 Abandoned: 18 ServiceLevel: 0 ServicelevelPerf: 0.0 Event: QueueMember Queue: test_queue  Location: Agent/101 Membership: static Penalty: 0 CallsTaken: 20 LastCall: 1118794338 Event: QueueMember Queue: test_queue  Location: Agent/102 Membership: static Penalty: 0 CallsTaken: 19 LastCall: 1118778909 Event: QueueMember Queue: test_queue  Location: Agent/103 Membership: static Penalty: 0 CallsTaken: 14 LastCall: 1118782495 Event: QueueMember Queue: test_queue  Location: Agent/104 Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Event: QueueMember Queue: test_queue  Location: Agent/105 Membership: sta
 tic
 Penalty: 0 CallsTaken: 9 LastCall: 1118779889 Event: QueueMember Queue: test_queue  Location: Agent/106 Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Event: QueueMember Queue: test_queue Location: Agent/107 Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Event: QueueMember Queue: test_queue Location: Agent/108 Membership: static Penalty: 0 CallsTaken: 146 LastCall: 1118795077 Event: QueueEntry Queue:  test_queue Position: 1 Channel: Zap/6-1 CallerID: 4042662907 Wait: 739 
Response: Goodbye Message: Thanks for all the fish. 
***
 
who knows how to get it through such info ,or there are other method for getting incoming calls number?
 
Any advice and help will be appreciated!

Best Regards,
Gary Li
		DO YOU YAHOO!? 
 
雅虎免费G邮箱－中国第一绝无垃圾邮件骚扰超大邮箱 ___
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Re: [Asterisk-Users] callqueues confused :(

[Steve Totaro]

t in your dial statement?

- Original Message - 
From: "Neil Bullock" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Friday, June 17, 2005 3:48 PM
Subject: [Asterisk-Users] callqueues confused :(


> > -- Started music on hold, class 'default', on
> > SIP/193.111.200.67-0815c790
> > -- outgoing agentcall, to agent '1001', on 'Local/[EMAIL PROTECTED],1'
> > -- Called Agent/1001
> > -- Executing Dial("Local/[EMAIL PROTECTED],2", "SIP/101|20|tr") in new
> > stack
> > -- Called 101
> > -- Agent/1001 is ringing
> > -- SIP/101-6fe1 is ringing
> > -- Agent/1001 is ringing
> > -- SIP/101-6fe1 answered Local/[EMAIL PROTECTED],2
> > -- Local/[EMAIL PROTECTED],1 answered, waiting for '#' to acknowledge
> > -- Started music on hold, class 'default', on Local/[EMAIL PROTECTED],2
> > -- Playing 'pbx-transfer' (language 'en')
> > -- Unable to find extension '' in context 'sip'
> > -- Playing 'pbx-invalid' (language 'en')
>
>
> I have a call queue that appears to work until the agent is requested to
> press #.
>
> At this point it tries to transfer the call but then says the extension
> is invalid.
>
> Any pointers appreciated, I've drawn a blank.
>
> Thanks
>
> Neil
>
>
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Re: [Asterisk-Users] tdm400p not working after cvs-head update

[Steve Totaro]

did you udate first?

  - Original Message - 
  From: 
  David Romero 
  
  To: Asterisk-Users@lists.digium.com 
  
  Sent: Friday, June 17, 2005 9:36 AM
  Subject: [Asterisk-Users] tdm400p not 
  working after cvs-head update
  I hava tdm400p card on [EMAIL PROTECTED] box, this card works fine for 
  weeks,today i did a CVS update to  the latest head files and the 
  card is not working.Zaptel Configuration== 
  Channel map:Channel 01: FXS Kewlstart (Default) (Slaves: 
  01)Channel 02: FXS Kewlstart (Default) (Slaves: 02)Channel 03: FXS 
  Kewlstart (Default) (Slaves: 03)Channel 04: FXS Kewlstart (Default) 
  (Slaves: 04)4 channels configured.ZT_CHANCONFIG failed on channel 1: 
  No such device or address (6) HELP!.thanks David 
  Romero## 
  
  

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Re: SV: [Asterisk-Users] Presence and IM?

[rsenykoff]

> 
> We have been running Asterisk for about a month now and one of the 
> things I miss the most is the ability to se who’s online and 
> available and who’s not. Surely, there’s the manager interface, but 
> unless you’d want to install extra software on each client computer,
> this is not a good option.
> 
> Then there’s the presence / IM function in SIP. Since we’re only 
> using SIP clients, this could easily solve some of our problems. 
> However, I cannot get this to work with Asterisk using Eyebeam. Is 
> this because the function is simply not supported within Asterisk?
> 
> If lack of support is the case, anyone knows if this feature is to 
> be implemented in the near future?
> 

We use Polycom IP500s which when used with a 'hint' in extensions.conf, 
can show presence via the 'buddy list.'

-Ron
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[Asterisk-Users] ASTCC Rate Calculation

[Darren Wiebe]

Good Day

Has anybody here looked closely at the call cost calculation in ASTCC?  
Can you duplicate the way the cost of a call is calculated?  I believe 
that there is an error in the code.  I have fixed it, I think and 
submitted a patch but we need user comments.  I would appreciate if 
anybody involved would slip over to chech out this link on the 
bugtracker and provide feedback. http://bugs.digium.com/view.php?id=4480
I may well be wrong but I believe the issue needs visiting.  Somebody 
was asking me how it calculates costs as they thought they knew what a 
call should cost.  I said "I'll show you".  Mistake, I could not come up 
with an answer that made sense.


Please let me know,

Darren Wiebe
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[Asterisk-Users] Asterisk ael files

[Shidan]

Hi noticing the cvs updates of late, I'm wondering if there is support
for fifo/shell commands in the extended dialplan language? can it
fully replace agi scripts? Looks really interesting...
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RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?

[Darren Wright]

 

 

 

You find me a reliable Teir 1 ISP T1 in New Hope, PA
for $300 to $400 





and I'll give you the amount I save over the next quarter. NPA-NXX is 





215-862. Good luck. 







 

I’ve got a Full T1 from a rather
large Mid-Atlantic CLEC for $291.  I’ve got about ½ dozen of them from DC
to Trenton, NJ.

 

-Darren

 








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RE: [Asterisk-Users] Bill seconds

[Terry H. Gilsenan]

 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of C F
> Sent: Saturday, 18 June 2005 2:20 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Bill seconds
> 
> > > cheapest in the world.
> > 
> > Ha Ha Ha Ha
> > 
> > How is your electricity sold? Hour, Watt, or by unit (KW/h)?
> > 
> > And as for cell phones being cheap, you have a receiver pays setup! 
> > How good is that, then you have so many competing Telcos that 
> > sometimes you just cannot call the house across the street without 
> > tracversing 3 exchanges, and then you have so many area 
> codes (each of 
> > which is so small) that almost all your calls are STD, then as for 
> > cheap, how is AUD$49.00/Month for all you can eat (that's about 
> > US$30/month) and all incoming calls are free, YES FREE! :)
> 
> Really???
> http://www.cucumber.com/fullinternational29.htm
> Look at the difference they (and everybody else) charges to 
> call the cell network. Check your bill (your landline bill) 
> next month and tell hoe much cell phone costs you.

The point I was making is that the charges are NOT on _My_ cell phone bill,
when I don't originate the call, however in .us if you get called, you pay,
that can easily cost you a heap of money that you can only control by
switching the phone off, and where is the point in that?

So if I rec'v 500 calls a week on my cell phone, it still costs me nothing.
And in some cases if I have the Cell and the Landline from the same telco
(in .au), calls between them are free too, regardless of where I happen to
be in australia at the time.

Oh, and cucumber seem to be doing you no favours either

I can place a call to the US using my Cell phone for 1-2c/minute, 
Caviat Emptor?

>  
> > So much for the "american"(sic) always making things better.
> > 
> 
> I knew that none americans might not see this unless pointed 
> out to them :)

Ha Ha Ha

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[Asterisk-Users] callqueues confused :(

[Neil Bullock]

> -- Started music on hold, class 'default', on
> SIP/193.111.200.67-0815c790
> -- outgoing agentcall, to agent '1001', on 'Local/[EMAIL PROTECTED],1'
> -- Called Agent/1001
> -- Executing Dial("Local/[EMAIL PROTECTED],2", "SIP/101|20|tr") in new
> stack
> -- Called 101
> -- Agent/1001 is ringing
> -- SIP/101-6fe1 is ringing
> -- Agent/1001 is ringing
> -- SIP/101-6fe1 answered Local/[EMAIL PROTECTED],2
> -- Local/[EMAIL PROTECTED],1 answered, waiting for '#' to acknowledge
> -- Started music on hold, class 'default', on Local/[EMAIL PROTECTED],2
> -- Playing 'pbx-transfer' (language 'en')
> -- Unable to find extension '' in context 'sip'
> -- Playing 'pbx-invalid' (language 'en')


I have a call queue that appears to work until the agent is requested to
press #.

At this point it tries to transfer the call but then says the extension
is invalid.

Any pointers appreciated, I've drawn a blank.

Thanks

Neil


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Re: [Asterisk-Users] Re: tdm400p not working after cvs-head update

[Soner Tari]

It looks from here like you've rebooted the system after checkout, but your 
system was not configured to load zaptel drivers at boot time.


Have you forgot to do 'make config' while in /usr/src/zaptel ?

Hope this helps

- Original Message - 
From: "David Romero" <[EMAIL PROTECTED]>

To: 
Sent: Friday, June 17, 2005 11:30 PM
Subject: [Asterisk-Users] Re: tdm400p not working after cvs-head update


I have tdm400p card on [EMAIL PROTECTED] box, this card works fine for weeks,

today i did a CVS update to the latest head files and the card is not
working.


Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)
4 channels configured.
ZT_CHANCONFIG failed on channel 1: No such device or address (6)

HELP!.

thanks

--
David Romero
##







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[Asterisk-Users] MGCP files for Polycom

[Rick Baranowski]

Title: MGCP files for Polycom






Does anybody know were I can download the MGCP files for the Polycom IP500?

Thanks



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Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?

[Robert Goodyear]

On Jun 17, 2005, at 7:56 AM, Daryl G. Jurbala wrote:
You find me a reliable Teir 1 ISP T1 in New Hope, PA for $300 to $400
and I'll give you the amount I save over the next quarter.  NPA-NXX is
215-862.  Good luck.

That sounds almost like Xeno's Paradox there... if you gave away the savings you still be paying the same amount thus half the savings would be...?

Sorry, just had to inject some Friday afternoon humor onto the list.

Seriously though, I was never able to get a T1 for that price anywhere myself until I moved to Orange County, CA.

-Rob.




-- 

Robert Goodyear  |  Managing Partner  |  Brand Up LLC
901 Calle Amanecer  |  Suite 150  |  San Clemente, CA 92673
Tel: 949/468.0370 x501  |  Fax: 949/468.0371  |  Cell/SMS: 949/981.7301
http://brand-up.com  |  [EMAIL PROTECTED]___
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[Asterisk-Users] Re: tdm400p not working after cvs-head update

[David Romero]

I have tdm400p card on [EMAIL PROTECTED] box, this card works fine for weeks,



today i did a CVS update to  the latest head files and the card is not working.





Zaptel Configuration

== 

Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)

Channel 02: FXS Kewlstart (Default) (Slaves: 02)

Channel 03: FXS Kewlstart (Default) (Slaves: 03)

Channel 04: FXS Kewlstart (Default) (Slaves: 04)

4 channels configured.

ZT_CHANCONFIG failed on channel 1: No such device or address (6)

 

HELP!.



thanks -- David Romero##
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Re: [Asterisk-Users] G729

[Carlos Chavez]

On Fri, 2005-06-17 at 12:33 -0400, David wrote:
> Hi All,
> 
>  
> 
> I have configured Line1 (2011) and Line2 (2012) in Sipura SPA-2000
> (latest Firmware) to use G729. In sip.conf I have set disallow=all,
> allow=g729
> 

Please take the time to read the Sipura documentation where it states
that you can only do ONE G729 call at a time on a SPA 1000, 2000 and
3000.  The processor in the unit is not powerful enough to do 2 G729
calls.  You have to allow ulaw or alaw so the other line can make a call
while the first is busy.


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[Asterisk-Users] Multiple phones on a Zap FXS extension

[Kelly Opal]

Hi
I have Asterisk up and running perfect with a Digium TDM400P card and 4
FXS ports. There are 4 AT&T 4-Line 954 phones hooked to the system Each
of the 4 lines is hooked to each phone. The problem is when you are on
line 1 (or any line) and someone else picks up line 1 they can here the
conversation. When the phone are hooked to the PSTN in the same way line
1 will be lite up to show that it is in use and you cannot join the call
without pressing a certain key sequence. How can I get the phones to act
like they do when they are connected to the PSTN when hooked to the
Asterisk server.

Any help is appreciated.

Kelly 

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Re: SV: [Asterisk-Users] Presence and IM?

[Time Bandit]

>  Take a look at the Asterisk Management Portal at
> http://sourceforge.net/projects/amportal
>  
>  It has a flash-based panel that will give you what you are looking for.

No need to install AMP to get this, just install FOP : http://www.asternic.org/

hth
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[Asterisk-Users] Phone lookup

[Aaron Daniel]

Quick question about call routing.  We're currently setting up our 
system so that any phone calls made from our system over a t1 line to 
another legacy system go through a dedicated t1 server.  Is there any 
method of checking to see if a number dialed exists on the system?  Any 
help would be appreciated.


Aaron Daniel
Sr. Voice Analyst
Sam Houston State University
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Re: SV: [Asterisk-Users] Presence and IM?

[Bryan M. Johns]

Bjorn,

Take a look at the Asterisk Management Portal at http://sourceforge.net/projects/amportal

It has a flash-based panel that will give you what you are looking for.




Bryan M. Johns
One Ring Networks
300 West Wieuca Road, NE
Building One
Suite 205
Atlanta, GA 30342
404.303.9900 x: 104
http://www.oneringnetworks.com




On Fri, 2005-06-17 at 20:33 +0200, Bjorn wrote:

Maybe, but that would not have been a reliable way of handling it, as not all users would necessarily use voicemail. Besides, I would think that this feature is supported by several SIP devices (it has to do with messaging), so it would be better If Asterisk supported this feature “by default”, no hacking needed.

 

Regards,

Bjorn 

 








Fra:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Dean Collins
Sendt: 17. juni 2005 18:11
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: RE: [Asterisk-Users] Presence and IM?




 

Hi Bjorn,

Maybe it could be done as some form of check against call forward to voicemail etc.

 

Dean

 








From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bjorn
Sent: Friday, 17 June 2005 11:51 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Presence and IM?




 

We have been running Asterisk for about a month now and one of the things I miss the most is the ability to se who’s online and available and who’s not. Surely, there’s the manager interface, but unless you’d want to install extra software on each client computer, this is not a good option.

 

Then there’s the presence / IM function in SIP. Since we’re only using SIP clients, this could easily solve some of our problems. However, I cannot get this to work with Asterisk using Eyebeam. Is this because the function is simply not supported within Asterisk?

 

If lack of support is the case, anyone knows if this feature is to be implemented in the near future?

 

Regards,

Bjorn 





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Re: SV: [Asterisk-Users] Presence and IM?

[Bryan M. Johns]

Bjorn,

Take a look at the Asterisk Management Portal at http://sourceforge.net/projects/amportal

It has a flash-based panel that will give you what you are looking for.




Bryan M. Johns
One Ring Networks
300 West Wieuca Road, NE
Building One
Suite 205
Atlanta, GA 30342
404.303.9900 x: 104
http://www.oneringnetworks.com




On Fri, 2005-06-17 at 20:33 +0200, Bjorn wrote:

Maybe, but that would not have been a reliable way of handling it, as not all users would necessarily use voicemail. Besides, I would think that this feature is supported by several SIP devices (it has to do with messaging), so it would be better If Asterisk supported this feature “by default”, no hacking needed.

 

Regards,

Bjorn 

 








Fra:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Dean Collins
Sendt: 17. juni 2005 18:11
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: RE: [Asterisk-Users] Presence and IM?




 

Hi Bjorn,

Maybe it could be done as some form of check against call forward to voicemail etc.

 

Dean

 








From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bjorn
Sent: Friday, 17 June 2005 11:51 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Presence and IM?




 

We have been running Asterisk for about a month now and one of the things I miss the most is the ability to se who’s online and available and who’s not. Surely, there’s the manager interface, but unless you’d want to install extra software on each client computer, this is not a good option.

 

Then there’s the presence / IM function in SIP. Since we’re only using SIP clients, this could easily solve some of our problems. However, I cannot get this to work with Asterisk using Eyebeam. Is this because the function is simply not supported within Asterisk?

 

If lack of support is the case, anyone knows if this feature is to be implemented in the near future?

 

Regards,

Bjorn 





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[Asterisk-Users] Spandsp - fax problem

[miguel]

I only get the first 1cm of the page in tiff.

Already tried to change the version of libtiff (.71), spandsp (pre18),
asterisk (CVS) and nothing!

The quality of image on that small band (1cm) is perfect.

Miguel


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[Asterisk-Users] PLEASE HELP X100P no responding

[Christian Callejon]

[EMAIL PROTECTED] ~]# modprobe zaptel[EMAIL PROTECTED] 
~]# modprobe wcfxoZT_CHANCONFIG failed on channel 1: No such device or 
address (6)FATAL: Error running install command for wcfxo
 
 
 
[EMAIL PROTECTED] ~]# ztcfg -vvv
 
Zaptel 
Configuration==Channel map:Channel 01: FXS 
Kewlstart (Default) (Slaves: 01)1 channels configured.ZT_CHANCONFIG 
failed on channel 1: No such device or address (6)
 
[EMAIL PROTECTED] /dev/zap]# ls -latotal 
0drwxr-xr-x  2 root 
root  120 jun 17 15:45 
.drwxr-xr-x  9 root 
root 5440 jun 17 15:45 
..crw-rw  1 asterisk asterisk 196, 254 jun 17 15:45 
channelcrw-rw  1 asterisk asterisk 196,   0 jun 17 15:45 
ctlcrw-rw  1 asterisk asterisk 196, 255 jun 17 15:45 
pseudocrw-rw  1 asterisk asterisk 196, 253 jun 17 15:45 
timer
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SV: [Asterisk-Users] Presence and IM?

[Bjorn]

Maybe, but that would not
have been a reliable way of handling it, as not all users would necessarily use
voicemail. Besides, I would think that this feature is supported by several SIP
devices (it has to do with messaging), so it would be better If Asterisk
supported this feature “by default”, no hacking needed.

 

Regards,

Bjorn 

 









Fra:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] På vegne av Dean Collins
Sendt: 17. juni 2005 18:11
Til: Asterisk Users Mailing List -
Non-Commercial Discussion
Emne: RE: [Asterisk-Users]
Presence and IM?



 

Hi Bjorn,

Maybe it could be done as
some form of check against call forward to voicemail etc.

 

Dean

 











From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Bjorn
Sent: Friday, 17 June 2005 11:51
AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Presence
and IM?



 

We have been running Asterisk for about a month now
and one of the things I miss the most is the ability to se who’s online
and available and who’s not. Surely, there’s the manager interface,
but unless you’d want to install extra software on each client computer,
this is not a good option.

 

Then there’s the presence / IM function in SIP.
Since we’re only using SIP clients, this could easily solve some of our
problems. However, I cannot get this to work with Asterisk using Eyebeam. Is
this because the function is simply not supported within Asterisk?

 

If lack of support is the case, anyone knows if this
feature is to be implemented in the near future?

 

Regards,

Bjorn 








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[Asterisk-Users] Phantom problem authenticating IAX2 with RSA

[Jon Lewis]

I'm getting exactly the same behavior as was posted about in
http://lists.digium.com/pipermail/asterisk-users/2004-March/040380.html

I've upgraded (both ends) to CVS stable (CVS-v1-0-06/17/05-13:15:49).

Jun 17 13:46:17 NOTICE[15942]: chan_iax2.c:4053 authenticate: No way to
send secret to peer 'a.b.c.d' (their methods: 4)

Immediately after that, I'll see frames go by with
Tx-Frame Retry[000] Subclass: NEW
Rx-Frame Retry[ No] Subclass: AUTHREQ
Tx-Frame Retry[000] Subclass: AUTHREP
Rx-Frame Retry[ No] Subclass: ACCEPT
that make it look very much like rsa authentication is being done, and the
call is accepted.

I noticed this while cleaning up my IAX config...moving away from
type=friend entries to a type=user and a type=peer entry for each system I
send/receive calls to/from.

i.e. on the remote end, I have:

[my.system.name]
username=my.system.name
type=user
auth=rsa
inkeys=my.system.name
context=my.system.name-iax
qualify=no
disallow=all
allow=g729
allow=gsm
deny=0.0.0.0/0.0.0.0
permit=[IP of my.system.name]

On the end I'm calling from:

[remote.system.name]
type=peer
username=my.system.name
auth=rsa
outkey=my.system.name
qualify=no
disallow=all
allow=g729
allow=gsm
host=remote.system.name

The test call is dialed as IAX2/remote.system.name/${EXTEN}
Is there a problem with my config, or is this just an iax2 cosmetic bug?
Each end does have appropriate rsa keys (readable by asterisk) in
/var/lib/asterisk/keys.

BTW, if I'm reading the docs correctly, there are multiple errors in the
wiki:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20IAX%20authentication#comments
where "allow" is incorrectly used [in the context of allowing an IP] where
"permit" was meant.

--
 Jon Lewis   |  I route
 Senior Network Engineer |  therefore you are
 Atlantic Net|
_ http://www.lewis.org/~jlewis/pgp for PGP public key_
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[Asterisk-Users] RE:Calculating the lenght of time in a call queue?

[Shad Mortazavi]

I don't get a queue_log file?

At what stage was this introduced?

Thanks

Shad
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[Asterisk-Users] PIX Firewall Ports and Access-Lists

[Geoff Manning]

Hello,

I am not too familiar with the settings in our PIX (learning though).

Here is the only access-list setting that we have in place for Asterisk:

access-list acl-prod permit udp any host EXTERNAL_*_IP_HERE eq 5060

In rtp.conf we are allowing ports 1 - 2.

We are not using SIP Fixup in our PIX due to firmware version.

How do I go about adding the ability for udp ports 1 - 2 to forward
to our Asterisk server?

We have intermittent audio issues on calls and I have narrowed it down
(hopefully) to the PIX.

Thanks!
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Re: [Asterisk-Users] Dell PowerEdge + TDM

[David Hajek]

Do you have analog TDM in it?

-David

Oswaldo Arratia wrote:


I bought a Dell SC1425 and installed a T1/E1 card from Digium and I tried to
configure it using [EMAIL PROTECTED] scripts and did not work, so I went the 
long way and
configure with zaptel's instructions and voila! It works like a charm.

Oswaldo 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Hajek
Sent: Friday, June 17, 2005 8:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Dell PowerEdge + TDM

Hi,

what new Dell servers are compatible and KNOWN to work with Digium TDM
cards? I've looked at Digium's compatibility list at
http://www.digium.com/index.php?menu=compatibility. Does this mean that
other Dell servers like SC1420, SC1425, 800, 1800 are working just fine with
TDM cards?

Can someone clarify this?

Thanks

-David

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Re: [Asterisk-Users] Can't switch span to E1-mode

[izo]

What card do you have ? Is there are jumper setup that you can specify
E1 or T1 ?
E1 cards a shipped set up as T1 by default


regards
m.
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[Asterisk-Users] txfax 18Kb file problem

[Vladyslav]

 Hi ALL.
 I have a problem with TxFax application. (RxFax is working properly)
 Txfax does not work when sending tiff files bigger than approx. 18Kb. If tiff
file smaller than 18Kb everything is OK. Tested in LAN with Panasonic UF-E1 fax
machine (tiff files was created by the rxfax from that machine) and SAP-3000.
Also tested with some remote fax machines (connected to PSTN and via VoIP).
Results are same. Small file in most cases is OK, bigger is FAILED.
 Environment: Fedora Core 2 and 3, spandsp 0.0.2pre18, libtiff 3.5.7-20.2 and
3.7.1-6, Asterisk 20050427CVS


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Re: [Asterisk-Users] bristuff-0.2.0-RC8g: zaptel error in suse 9.2

[Marco Parmeggiani]

Manuel Casal ha scritto:


make[1]: Entering directory `/usr/src/linux-2.6.8-24.16-obj/i386/smp'
make[1]: *** No rule to make target `modules'.  Stop.
make[1]: Leaving directory `/usr/src/linux-2.6.8-24.16-obj/i386/smp'
make: *** [linux26] Error 2
linux:/usr/src/asterisk/bristuff-0.2.0-RC8g/zaptel-1.0.7 #

Now what?:(



i'm using a Debian.
i'm missing those *-obj links in my /usr/src

drwxr-xr-x  19 root root 4096 Jun 17 18:04 kernel-source-2.6.11
lrwxrwxrwx   1 root src18 May 24 14:36 linux -> /usr/src/linux-2.6
lrwxrwxrwx   1 root src20 May 24 13:54 linux-2.6 -> 
kernel-source-2.6.11


and it compiles fine.

HTH
ciao
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Re: [Asterisk-Users] bristuff-0.2.0-RC8g: zaptel error in suse 9.2

[Manuel Casal]

Marco Parmeggiani escribió:


Manuel Casal ha scritto:


I made the "make menuconfig" and "make dep" in the kernel sources.



i do not remember well how i solved that problem but i'm sure that 
"make dep" will issue you a warning and stop.
run "make" to start the kernel build process and then stop it after 
few seconds. it will create the necessary symlinks in the kernel tree.

maybe there's a more elegant solution but this should work.

ciao
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I did that and this is the result... a new kind of error


linux:/usr/src/asterisk/bristuff-0.2.0-RC8g/zaptel-1.0.7 # make all
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o gendigits.o gendigits.c

cc -o gendigits gendigits.o -lm
./gendigits
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"makefw.c   -o makefw

./makefw tormenta2.rbt tor2fw > tor2fw.h
Loaded 69900 bytes from file
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o ztcfg.o ztcfg.c
cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DBUILDING_TONEZONE -o zonedata.lo 
zonedata.c
cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DBUILDING_TONEZONE -o tonezone.lo 
tonezone.c

ar rcs libtonezone.a zonedata.lo tonezone.lo
cc -o ztcfg ztcfg.o -lm -L. libtonezone.a
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o torisatool.o torisatool.c

cc -o torisatool torisatool.o
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o ztmonitor.o ztmonitor.c

cc -o ztmonitor ztmonitor.o
cc -c ztspeed.c
cc -o ztspeed ztspeed.o
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o zttool.o zttool.c

cc -o zttool zttool.o -lnewt
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"zttest.c   -o zttest
make -C /lib/modules/`uname -r`/build 
SUBDIRS=/usr/src/asterisk/bristuff-0.2.0-RC8g/zaptel-1.0.7 modules

make[1]: Entering directory `/usr/src/linux-2.6.8-24.16-obj/i386/smp'
make[1]: *** No rule to make target `modules'.  Stop.
make[1]: Leaving directory `/usr/src/linux-2.6.8-24.16-obj/i386/smp'
make: *** [linux26] Error 2
linux:/usr/src/asterisk/bristuff-0.2.0-RC8g/zaptel-1.0.7 #

Now what?:(


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Re: [Asterisk-Users] G729

[Erick Weber V.]

Title: Untitled Document



Hi,
 
The Sipura SPA-2000 can only support one G729 
call
 
Regards
 
Erick

  - Original Message - 
  From: 
  David 
  To: 'Asterisk Users Mailing List - 
  Non-Commercial Discussion' 
  Sent: Friday, June 17, 2005 11:33 
AM
  Subject: [Asterisk-Users] G729
  
  
  Hi All,
   
  I have configured Line1 (2011) and 
  Line2 (2012) in Sipura SPA-2000 (latest Firmware) to use 
  G729. In sip.conf I have set disallow=all, allow=g729
   
  If Line1 is in use by an agent, then Line2 won't 
  work and vice versa (Inbound Calls Only).  I have 40 license for G729. so there shouldn't be any 
  issue with the license. 
   
  I'm getting the following error 
  msg:
   
   -- Called 2012    -- Got SIP 
  response 488 "Not Acceptable Here" back from 192.168.10.103  == No 
  one is available to answer at this time (1:0/0/0)  == Auto 
  fallthrough, channel 'IAX2/[EMAIL PROTECTED]:4569-5' 
  status is 'NOANSWER'    -- Hungup 'IAX2/[EMAIL PROTECTED]:4569-5'
   
  If I change 2012 to ULAW, it works fine. It seems 
  that I can't have two lines configured as a G729. 
   
  Do you guys have any idea why this 
  happening?
  Regards, 
   
   
  
  

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Re: [Asterisk-Users] Bill seconds

[Jean-Michel Hiver]

If you tell an american 'this is it', s/he will think of ways to change it and 
make
it another way, thats why we have a '96 telecommunications act, and
why having cell phones in the states are the cheapest in the world.
 

Oh yeah. Americans are always faster, better, bigger. We cheese eating 
surrending monkeys get the idea ;-)


Can we go back to asterisk stuff now?

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Re: [Asterisk-Users] G729

[Bruce Komito]

The Sipura SPA2000 only supports one G729 call at a time.  Same with the
Linksys PAP2.

Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815


On Fri, 17 Jun 2005, David wrote:

> Hi All,
>
>
>
> I have configured Line1 (2011) and Line2 (2012) in Sipura SPA-2000 (latest
> Firmware) to use G729. In sip.conf I have set disallow=all, allow=g729
>
>
>
> If Line1 is in use by an agent, then Line2 won't work and vice versa
> (Inbound Calls Only).  I have 40 license for G729. so there shouldn't be any
> issue with the license.
>
>
>
> I'm getting the following error msg:
>
>
>
>  -- Called 2012
> -- Got SIP response 488 "Not Acceptable Here" back from 192.168.10.103
>   == No one is available to answer at this time (1:0/0/0)
>   == Auto fallthrough, channel 'IAX2/[EMAIL PROTECTED]:4569-5' status is
> 'NOANSWER'
> -- Hungup 'IAX2/[EMAIL PROTECTED]:4569-5'
>
>
>
>
> If I change 2012 to ULAW, it works fine. It seems that I can't have two
> lines configured as a G729.
>
>
>
> Do you guys have any idea why this happening?
>
>
> Regards,
>
>
>
>
>
>
>
> This message has been categorized as "Legitimate" by Bayesian Analyzer.
> If you do not agree, please click on the link below to train the Analyzer.
> http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-06-17%5Cd81c0f432a8146fd9b6064a4b2fc65b8&C=2
>
> --
> ---
> This message has been inspected by DynaComm i:mail
> ---
>

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[Asterisk-Users] SIP transfer/REFER to voicemail problem

[B Ayers]

For anyone else who might run into this, I got around the transferring to
voicemail problem by putting a "canreinvite=no" line into the section for each
caller's SIP address in sip.conf.  Not ideal, but it works.

I also had to add a "dtmfmode=inband" for my Mediatrix 1204 addresses to be
able to access the voicemail commands.

--
I've google for hours trying to find a discussion of a similar problem as the
one I'm having, so forgive me if this has come up before.  If it has, please
point me in the right direction!

The problem occurs when a caller (A) is transferred by an intermediary party
(B) to voicemail (Voicemail or VoicemailMain), either directly or by being
taken to voicemail when the callee (C) doesn't answer.  Caller (A) hears the
Asterisk voicemail prompts, but the voicemail application doesn't hear any
audio or DTMF.

Easy to duplicate:
1.) A -> B (INVITE)
2.) B -> C (REFER A to C)
3.) A -> C

More descriptive:
1.)  Caller (A) calls intermediary (B).  (B can be any SIP user agent)
2.)  Intermediary (B) REFERs caller (A) to callee (C)
3.)  C is either a SIP UA which times-out and Asterisk takes to Voicemail, or
an extension tied to VoicemailMain.

I've come across a thread saying that the Asterisk voicemail system only uses
the GSM codec, but if this were the problem, then how can the caller (using
mu-law) hear the voicemail prompts?  Would Asterisk be doing a half duplex
protocol conversion?

Any insight would be greatly appreciated!!


Current configuration:
Fedora Core 1
Asterisk - 1.0.7 (had same problem on 1.0.6)
SJPhone - 1.50.271d, Mar 11 2005  (WinXP)
XLite - 1103m build stamp 14262  (WinXP)
Zultys Zip2 - ZUTS 3.52


sip.conf exerpt:

[6003]  ; (A)
type=friend
regexten=6003
username=6003
host=dynamic
disallow=all
;allow=gsm
allow=ulaw

[6004]  ; (C)
type=friend
regexten=6004
username=6004
host=dynamic
disallow=all
;allow=gsm
allow=ulaw

[2101]  ; (B)
type=friend
regexten=2101
username=2101
host=dynamic
disallow=all
;allow=gsm
allow=ulaw


extensions.conf exerpt:

exten => 6003,1,Dial(SIP/1003,15)
exten => 6003,2,Voicemail(u1003)
exten => 6003,102,Voicemail(b1003)

exten => 6004,1,Dial(SIP/1004,5)
exten => 6004,2,Voicemail(u1004)
exten => 6004,102,Voicemail(b1004)

exten => 2101,1,Dial(SIP/2101)

exten => 8500,1,VoicemailMain
exten => 8500,2,Hangup


Asterisk (-dvvgc) with sip debug on (REFER-ing caller to VoicemailMain) :

   -- No username but # key pressed. Using CID '6003'
   -- Playing 'vm-password' (language 'en')
Urgent handler
   -- Incorrect password '' for user '6003' (context = ,any)
   -- Playing 'vm-incorrect-mailbox' (language 'en')
Urgent handler

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[Asterisk-Users] tdm400p not working after cvs-head update

[David Romero]

I hava tdm400p card on [EMAIL PROTECTED] box, this card works fine for weeks,

today i did a CVS update to  the latest head files and the card is not working.


Zaptel Configuration
== 
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)
4 channels configured.
ZT_CHANCONFIG failed on channel 1: No such device or address (6)
 
HELP!.

thanks 

 David Romero##
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[Asterisk-Users] G729

[David]

Title: Untitled Document




Hi All,
 
I have configured Line1 (2011) and 
Line2 (2012) in Sipura SPA-2000 (latest Firmware) to use 
G729. In sip.conf I have set disallow=all, allow=g729
 
If Line1 is in use by an agent, then Line2 won't 
work and vice versa (Inbound Calls Only).  I have 40 license for G729. so there shouldn't be any 
issue with the license. 
 
I'm getting the following error 
msg:
 
 -- Called 2012    -- Got SIP 
response 488 "Not Acceptable Here" back from 192.168.10.103  == No one 
is available to answer at this time (1:0/0/0)  == Auto fallthrough, 
channel 'IAX2/[EMAIL PROTECTED]:4569-5' 
status is 'NOANSWER'    -- Hungup 'IAX2/[EMAIL PROTECTED]:4569-5'
 
If I change 2012 to ULAW, it works fine. It seems that 
I can't have two lines configured as a G729. 
 
Do you guys have any idea why this 
happening?
Regards, 
 
 
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Re: [Asterisk-Users] Bill seconds

[C F]

On 6/16/05, Tony Hoyle <[EMAIL PROTECTED]> wrote:
> Terry H. Gilsenan wrote:
> 
> > And as for cell phones being cheap, you have a receiver pays setup! How good
> > is that, then you have so many competing Telcos that sometimes you just
> 
> I believe that's unique to the US, the idea of paying for actually
> receiving calls...  don't know why they stand for it but I guess the
> masses don't know any better.
> 
> Here in the UK mobile phones are only £15/month on the cheapest tariffs
> which equates to about $25..

$.30 a min to UK cell how is this cheap?
http://www.cucumber.com/fullinternational29.htm

 
> This thread could easily become a 'my phone is cheaper than yours'
> argument :)  I'm just waiting for someone from Japan to post... (they
> *are* the cheapest in the world I believe, by a long margin).

As for Japan I don't know you might be right.

> 
> Tony
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Re: [Asterisk-Users] Bill seconds

[C F]

> > cheapest in the world.
> 
> Ha Ha Ha Ha
> 
> How is your electricity sold? Hour, Watt, or by unit (KW/h)?
> 
> And as for cell phones being cheap, you have a receiver pays setup! How good
> is that, then you have so many competing Telcos that sometimes you just
> cannot call the house across the street without tracversing 3 exchanges, and
> then you have so many area codes (each of which is so small) that almost all
> your calls are STD, then as for cheap, how is AUD$49.00/Month for all you
> can eat (that's about US$30/month) and all incoming calls are free, YES
> FREE! :)

Really???
http://www.cucumber.com/fullinternational29.htm
Look at the difference they (and everybody else) charges to call the
cell network. Check your bill (your landline bill) next month and tell
hoe much cell phone costs you.
 
> So much for the "american"(sic) always making things better.
> 

I knew that none americans might not see this unless pointed out to them :)

> The best phone and postal services by far in in Australia. There are no
> peers at all.
> 
> Sorry, I just had to call that bluff
> 
> T
>
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[Asterisk-Users] Can't switch span to E1-mode

[Yousef Herzallah]

Hi,
This error I got it just when I gonfigure zaptel support isdneuro 31
channels.
But if I configure zaptel to support T1 and just 24 channels I have no
problem.


#
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
#
loadzone = it
span=1,1,0,ccs,hdb3,crc4
bchan=1-23
dchan=16
bchan=17-31

[EMAIL PROTECTED] etc]# cat /etc/asterisk/zapata.conf
;
; Zapata telephony interface
;
; Configuration file
;
; You need to restart Asterisk to re-configure the Zap channel
; CLI> reload chan_zap.so
;   will reload the configuration file,
;   but not all configuration options are
;   re-configured during a reload.
[channels]
signalling=pri_cpe
immediate=no
switchtype=EuroISDN
pridialplan=unknown
context=incoming
usecallerid=yes
group=1
channel => 1-23
channel => 17-31


[EMAIL PROTECTED] etc]# /etc/init.d/zaptel restart
Unloading zaptel hardware drivers: wcusb wctdm wcfxo wcte11xp wct1xxp
wct4xxp tor2.
Removing zaptel module:[  OK  ]
Loading zaptel framework:  [  OK  ]
Waiting for zap to come online...OK
Loading zaptel hardware modules:Running ztcfg:  ZT_CHANCONFIG failed on
channel 25: No such device or address (6)
   [FAILED]
[EMAIL PROTECTED] etc]# dmesg
TE110P: Setting up global serial parameters for T1 FALC V1.2
TE110P: Successfully initialized serial bus for card
Found a Wildcard: Digium Wildcard TE110P T1/E1

[EMAIL PROTECTED] etc]# ztcfg -v

Zaptel Configuration
==
SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)
31 channels configured.
ZT_CHANCONFIG failed on channel 25: No such device or address (6)



Now I'll show how it work with 24 channels

[EMAIL PROTECTED] etc]# dmesg
TE110P: Setting up global serial parameters for T1 FALC V1.2
TE110P: Successfully initialized serial bus for card
Found a Wildcard: Digium Wildcard TE110P T1/E1
Registered tone zone 11 (Italy)
TE110P: Span configured for ESF/B8ZS
Calling startup (flags is 4099)
wcte1xxp: Setting yellow alarm
TE110P: Span configured for ESF/B8ZS
Calling startup (flags is 4099)
Registered tone zone 11 (Italy)
TE110P: Span configured for ESF/B8ZS
Calling startup (flags is 4099)
Registered tone zone 11 (Italy)
usbcore: registered new driver wcusb
Wildcard USB FXS Interface driver registered
TE110P: Span configured for ESF/B8ZS
Calling startup (flags is 4099)
Registered tone zone 11 (Italy)
TE110P: Span configured for ESF/B8ZS
Calling startup (flags is 4099)
Registered tone zone 11 (Italy)

And no errors.
Help,

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RE: [Asterisk-Users] Presence and IM?

[Dean Collins]

Hi Bjorn,

Maybe it could be done as some form of
check against call forward to voicemail etc.

 

Dean

 











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bjorn
Sent: Friday, 17 June 2005 11:51
AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Presence
and IM?



 

We have been running Asterisk for about a month now
and one of the things I miss the most is the ability to se who’s online
and available and who’s not. Surely, there’s the manager interface,
but unless you’d want to install extra software on each client computer,
this is not a good option.

 

Then there’s the presence / IM function in SIP.
Since we’re only using SIP clients, this could easily solve some of our
problems. However, I cannot get this to work with Asterisk using Eyebeam. Is
this because the function is simply not supported within Asterisk?

 

If lack of support is the case, anyone knows if this
feature is to be implemented in the near future?

 

Regards,

Bjorn 








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Re: [Asterisk-Users] 2nd Dialtone after answer

[Henry Coleman]

Hi all, DID is Direct Inward Dial where the last 3 or 4 digits of the 
dialed number are passed through and are used/translated to call a 
specific extension. (See Centrex)
DISA is Direct System Access where incoming line(s)are auto-answered and 
receive internal dial tone, the caller then has access to the facilities 
of the system.(including calling an extension.)


I hope this clears things up

TTFN Henry



Chris Coulthurst wrote:

Check out DISA.

Chris Coulthurst
[EMAIL PROTECTED]
 



|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Oswaldo Arratia

|Sent: Friday, June 17, 2005 7:51 AM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: [Asterisk-Users] 2nd Dialtone after answer
|
|
|Hi
|I am trying to achive this for a specific need of a customer.
|
|He has a DID pointed to an Asterisk server, I need to provide 
|him dialtone when the calls hits the server. How can I achieve this?

|
|Let's say something like this:
|
|Exten => s,1,Answer
|Exten => s,2, "Provide Dial tone"
|Exten => s,3, "Dial the number the person will enter after 
|receiving the dial tone" Exten => s,4,Hangup

|
|Any ideas?
|
|Thanks very much
|
|Oswaldo
|
|
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Re: [Asterisk-Users] bristuff-0.2.0-RC8g: zaptel error in suse 9.2

[Marco Parmeggiani]

Manuel Casal ha scritto:


I made the "make menuconfig" and "make dep" in the kernel sources.


i do not remember well how i solved that problem but i'm sure that "make 
dep" will issue you a warning and stop.
run "make" to start the kernel build process and then stop it after few 
seconds. it will create the necessary symlinks in the kernel tree.

maybe there's a more elegant solution but this should work.

ciao
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[Asterisk-Users] Presence and IM?

[Bjorn]

We have been running Asterisk for about a month now
and one of the things I miss the most is the ability to se who’s online
and available and who’s not. Surely, there’s the manager interface,
but unless you’d want to install extra software on each client computer,
this is not a good option.

 

Then there’s the presence / IM function in SIP.
Since we’re only using SIP clients, this could easily solve some of our
problems. However, I cannot get this to work with Asterisk using Eyebeam. Is
this because the function is simply not supported within Asterisk?

 

If lack of support is the case, anyone knows if this
feature is to be implemented in the near future?

 

Regards,

Bjorn 






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RE: [Asterisk-Users] WiFi IP Phones

[trixter http://www.0xdecafbad.com]

Given that they are radio transmitters, there is always the risk that
they can cause a spark and ignite something.  Additionally, reports have
happened of the battery itself getting shorted when removed and causing
everything from bullets to other explosive situations to occur.  When
you short the metal contacts on the battery it gets very hot, that heat
can cause things to ignite (gasoline for example needs about 500 degrees
farenheight to ignite).

Cell phones are normally upto 600mW PEP (max FCC allowed for a
handheld).  Wifi devices are normally upto 200mW PEP.  Amplifiers can
change this.  ETSI in europe limits to 100mW PEP and in Japan wifi is
limited to 10mW or something silly.  Thus its less of a concern in other
regions.

If you put a little bit of metal in the microwave you will see sparks on
the metal.  This is because there is a difference in RF potential across
two points, and an arcing occurs.  Granted most microwaves are 600-1000W
PEP, or 1000+ times the power, but the same type of situation can occur
if conditions are right.

In short there is no way to completly reduce the chance of explosions of
certain substances, to get a cell phone far enough away to mitigate that
danger is a matter of inches (1 inch == 2.54 cm), a wifi device, having
less power, is an even shorter distance.  If you are very near dangerous
substances that could be set off this way you should (hopefully anyway)
be trained in proper procedure there.

There is more risk (I think anyway) of filling a plastic gas can inside
a pickup bed with a plastic liner (plastic on plastic can create a
static discharge).  Cigarettes often dont get hot enough to ignite
gasoline (outside movies) because only when inhaling do they near hot
enough, just tossing one onto gas its normally 50 degrees below the
flash point.

Remember liquids dont burn, only gasses do in normal physics anyway
(special conditions can occur with extreme temperatures and pressures).
Gas station tanks are grounded if metal, the pumps certainly are.  This
further mitigates risk.

AFAIK there arent regulations that require them to prevent explosions,
any regulations like that would be on the devices that contain or
transport such materials that are likely to explode.  Because people
transmit a lot of power on mobile radios, those working with detonators
at construction sites often are required to put out signs saying 'dont
transmit explosive danger' because you can cause a false fire signal to
be sent to the detonator if you kick out enough power, but even those
devices are typically shielded to minimize this risk.



On Fri, 2005-06-17 at 10:49 -0400, Daryl G. Jurbala wrote:
> > -Original Message-
> > From: [EMAIL PROTECTED] 
> > [mailto:[EMAIL PROTECTED] On Behalf Of 
> > Anton Krall
> > Sent: Wednesday, June 15, 2005 3:01 PM
> > To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> > Subject: [Asterisk-Users] WiFi IP Phones
> > 
> > Guys.
> > 
> > I know there are wifi sip phones out there but I have a 
> > question, are any of these phones "anti explosive"? By that I 
> > mean, there are certain regulations about phones or cel 
> > phones that are not recommended to operate in environments 
> > like gas stations due to sparks and the chance of ingiting gas fumes.
> 
> You are referring to (in the US anyway) certification as "intrinsically
> safe".
> 
> I don't know either way about phones listed as such, but with the right
> terminology you might have better liuck searching.
> 
>  voiceverified. | Daryl G. Jurbala
>  -- | Chief Technology Officer
> | 215.862.1160 x235 (Office)
> It had to be you!   | 215.862.9880 (FAX) 
> 
> 
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-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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Re: [Asterisk-Users] MFC/R2

[j_amorim]

The libsupertone library installation happened because the libxml2 library 
was not installed. After it was installed the problem was solved. 

Now I am experiencing the problem following: 

Do you have any tip??? 

Thanks. 

[chan_unicall.so] => (Unified call processing (UniCall)) 
  == Parsing '/etc/asterisk/unicall.conf': Found 
Loading protocol mfcr2 
Country 5 
-- Registered channel 9, mfcr2 signalling 
Country 5 
-- Registered channel 10, mfcr2 signalling 
Country 5 
-- Registered channel 11, mfcr2 signalling 
Country 5 
-- Registered channel 12, mfcr2 signalling 
Country 5 
-- Registered channel 13, mfcr2 signalling 
Country 5 
-- Registered channel 14, mfcr2 signalling 
Country 5 
-- Registered channel 15, mfcr2 signalling 
Country 5 
-- Registered channel 16, mfcr2 signalling 
Country 5 
-- Registered channel 17, mfcr2 signalling 
Country 5 
-- Registered channel 18, mfcr2 signalling 
Country 5 
-- Registered channel 19, mfcr2 signalling 
Country 5 
-- Registered channel 20, mfcr2 signalling 
Country 5 
-- Registered channel 21, mfcr2 signalling 
Country 5 
-- Registered channel 22, mfcr2 signalling 
Country 5 
-- Registered channel 23, mfcr2 signalling 
Country 5 
-- Registered channel 25, mfcr2 signalling 
Country 5 
-- Registered channel 26, mfcr2 signalling 
Country 5 
-- Registered channel 27, mfcr2 signalling 
Country 5 
-- Registered channel 28, mfcr2 signalling 
Country 5 
-- Registered channel 29, mfcr2 signalling 
Country 5 
-- Registered channel 30, mfcr2 signalling 
Country 5 
-- Registered channel 31, mfcr2 signalling 
Country 5 
-- Registered channel 32, mfcr2 signalling 
Country 5 
-- Registered channel 33, mfcr2 signalling 
Country 5 
-- Registered channel 34, mfcr2 signalling 
Country 5 
-- Registered channel 35, mfcr2 signalling 
Country 5 
-- Registered channel 36, mfcr2 signalling 
Country 5 
-- Registered channel 37, mfcr2 signalling 
Country 5 
-- Registered channel 38, mfcr2 signalling 
Country 5 
-- Registered channel 39, mfcr2 signalling 
Country 5 
-- Registered channel 40, mfcr2 signalling 
Country 5 
-- Registered channel 41, mfcr2 signalling 
Country 5 
-- Registered channel 42, mfcr2 signalling 
Country 5 
-- Registered channel 43, mfcr2 signalling 
Country 5 
-- Registered channel 44, mfcr2 signalling 
Country 5 
-- Registered channel 45, mfcr2 signalling 
Country 5 
-- Registered channel 46, mfcr2 signalling 
Country 5 
-- Registered channel 47, mfcr2 signalling 
Country 5 
-- Registered channel 48, mfcr2 signalling 
Country 5 
-- Registered channel 49, mfcr2 signalling 
Country 5 
-- Registered channel 50, mfcr2 signalling 
Country 5 
-- Registered channel 51, mfcr2 signalling 
Country 5 
-- Registered channel 52, mfcr2 signalling 
Country 5 
-- Registered channel 53, mfcr2 signalling 
Country 5 
-- Registered channel 54, mfcr2 signalling 
Country 5 
-- Registered channel 56, mfcr2 signalling 
Country 5 
-- Registered channel 57, mfcr2 signalling 
Country 5 
-- Registered channel 58, mfcr2 signalling 
Country 5 
-- Registered channel 59, mfcr2 signalling 
Country 5 
-- Registered channel 60, mfcr2 signalling 
Country 5 
-- Registered channel 61, mfcr2 signalling 
Country 5 
-- Registered channel 62, mfcr2 signalling 
Country 5 
-- Registered channel 63, mfcr2 signalling 
Country 5 
-- Registered channel 64, mfcr2 signalling 
Country 5 
-- Registered channel 65, mfcr2 signalling 
Country 5 
-- Registered channel 66, mfcr2 signalling 
Country 5 
-- Registered channel 67, mfcr2 signalling 
Country 5 
-- Registered channel 68, mfcr2 signalling 
Country 5 
-- Registered channel 69, mfcr2 signalling 
Country 5 
-- Registered channel 70, mfcr2 signalling 
Parsing tone set 
Hit dial-tone 
Step - Frequency=350.00+440.00 [1.00%]Level=-13.00+-13.00 Recognition 
length=0.30 [10.00%] 
>>>Detector element350440300  0 
Hit dial-tone 
Step - Cycles=3 
Step - Frequency=350.00+440.00 [1.00%]Level=-13.00+-13.00Length=0.10 
[10.00%] 
>>>Detector element350440 60140 
Step - Length=0.10 [10.00%] 
>>>Detector element  0  0 60140 
Step - Frequency=350.00+440.00 [1.00%]Level=-13.00+-13.00 Recognition 
length=0.30 [10.00%] 
>>>Detector element350440300  0 
Hit ringback-tone 
Step - Cycles=0 
Step - Frequency=440.00+480.00 [1.00%]Level=-13.00+-13.00Length=0.40 
[10.00%] 
>>>Detector element440480330470 
Step - Length=0.20 [10.00%] 
>>>Detector element  0  0150250 
Step - Frequency=440.00+480.00 [1.00%]Level=-13.00+-13.00Length=0.40 
[10.00%] 
>>>Detector element440480330470 
Step - Length=3.00 [10.00%] Recognition length=0.60 [10.00%] 

RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?

[Daryl G. Jurbala]

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Huddleston, Robert
> Sent: Tuesday, June 14, 2005 3:49 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
> 
> Anyone paying over $450 for a T1 is being ripped off...
> If you are in VA,MD,DC,PA,DE,NJ you can get an integrated 
> VoIP T1 for $300 - $400 and a flat internet t1 for about $400.
> The integrated VoIP T1 is great because it's handed off as an 
> ethernet - no need for a csu/dsu 

Ummm...no.  Maybe if you are in or very near a city you can, but not
everywhere.

You find me a reliable Teir 1 ISP T1 in New Hope, PA for $300 to $400
and I'll give you the amount I save over the next quarter.  NPA-NXX is
215-862.  Good luck.

 voiceverified. | Daryl G. Jurbala
 -- | Chief Technology Officer
| 215.862.1160 x235 (Office)
It had to be you!   | 215.862.9880 (FAX) 


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RE: [Asterisk-Users] WiFi IP Phones

[Daryl G. Jurbala]

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Dean Collins
> Sent: Thursday, June 16, 2005 9:57 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] WiFi IP Phones
> 
> Ahmm Andrew, are you sure they are steel?
> 
> It's been a long time since I did any work in this space but 
> we used to install them in plastic not metal.plastic 
> works better with the radio waves.

IS does not necessarily mean steel.  My Motorola alpha pager, and my
Motorola XTS3000 radio are both plastic and IS listed.


 voiceverified. | Daryl G. Jurbala
 -- | Chief Technology Officer
| 215.862.1160 x235 (Office)
It had to be you!   | 215.862.9880 (FAX) 


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RE: [Asterisk-Users] Calculating the lenght of time in a call queue?

[Niall McCarthy]

Time waiting for an agent is one of the fields recorded in the queue_log
see the following 
http://voip-info.org/tiki-index.php?page=Asterisk%20log%20queue_log

Regards,
Mac.



-Original Message-
From: Shad Mortazavi [mailto:[EMAIL PROTECTED] 
Sent: 17 June 2005 15:54
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Calculating the lenght of time in a call
queue?


Dear All,

I'm running version 0.7.1 of Asterisk server for our global help desk.

We have put together a comprehensive reporting package for static's from
the CDR. 

I'm not able to calculate the time a call is in the queue before it goes
to an agent? 

I would appreciate help with working this out.

Warm Regards and Thanks

Shad Mortazavi
--
Nexus Global Technical Manager
n|m Nexus Management Inc 

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Re: [Asterisk-Users] 2nd Dialtone after answer

[David John Walsh]

it sounds like you need to investigate the application called DISA..
it might be what you are looking for

if not, what dial tone is your client expecting? (internal, external - other??)

David

On 17/06/05, Oswaldo Arratia <[EMAIL PROTECTED]> wrote:
> Hi
> I am trying to achive this for a specific need of a customer.
> 
> He has a DID pointed to an Asterisk server, I need to provide him dialtone
> when the calls hits the server. How can I achieve this?
> 
> Let's say something like this:
> 
> Exten => s,1,Answer
> Exten => s,2, "Provide Dial tone"
> Exten => s,3, "Dial the number the person will enter after receiving the
> dial tone"
> Exten => s,4,Hangup
> 
> Any ideas?
> 
> Thanks very much
> 
> Oswaldo
> 
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RE: [Asterisk-Users] WiFi IP Phones

[Daryl G. Jurbala]

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Anton Krall
> Sent: Wednesday, June 15, 2005 3:01 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [Asterisk-Users] WiFi IP Phones
> 
> Guys.
> 
> I know there are wifi sip phones out there but I have a 
> question, are any of these phones "anti explosive"? By that I 
> mean, there are certain regulations about phones or cel 
> phones that are not recommended to operate in environments 
> like gas stations due to sparks and the chance of ingiting gas fumes.

You are referring to (in the US anyway) certification as "intrinsically
safe".

I don't know either way about phones listed as such, but with the right
terminology you might have better liuck searching.

 voiceverified. | Daryl G. Jurbala
 -- | Chief Technology Officer
| 215.862.1160 x235 (Office)
It had to be you!   | 215.862.9880 (FAX) 


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[Asterisk-Users] Programinng Aplication with Music on Hold

[Jose Raul Pineda Lemus]

Hi,

I am programing  one aplications to hear my e-mail on PBX asterisk. I
want to have music on hold when the my aplications get the e-mail the 
mail server real.
I am doing un IVR to this aplicationes.

How I can do that?
When is the routines on the source asterisk  to music on hold?
What files I can see, for one example?


Thank,

Raul Pineda
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RE: [Asterisk-Users] 2nd Dialtone after answer

[Chris Coulthurst]

Check out DISA.

Chris Coulthurst
[EMAIL PROTECTED]
 


|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Oswaldo Arratia
|Sent: Friday, June 17, 2005 7:51 AM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: [Asterisk-Users] 2nd Dialtone after answer
|
|
|Hi
|I am trying to achive this for a specific need of a customer.
|
|He has a DID pointed to an Asterisk server, I need to provide 
|him dialtone when the calls hits the server. How can I achieve this?
|
|Let's say something like this:
|
|Exten => s,1,Answer
|Exten => s,2, "Provide Dial tone"
|Exten => s,3, "Dial the number the person will enter after 
|receiving the dial tone" Exten => s,4,Hangup
|
|Any ideas?
|
|Thanks very much
|
|Oswaldo
|
|
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|http://lists.digium.com/mailman/listinfo/asteri|sk-users
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|UNSUBSCRIBE or update options visit:
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[Asterisk-Users] Calculating the lenght of time in a call queue?

[Shad Mortazavi]

Dear All,

I'm running version 0.7.1 of Asterisk server for our global help desk.

We have put together a comprehensive reporting package for static's from
the CDR. 

I'm not able to calculate the time a call is in the queue before it goes
to an agent? 

I would appreciate help with working this out.

Warm Regards and Thanks

Shad Mortazavi
--
Nexus Global Technical Manager
n|m Nexus Management Inc 

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[Asterisk-Users] 2nd Dialtone after answer

[Oswaldo Arratia]

Hi
I am trying to achive this for a specific need of a customer.

He has a DID pointed to an Asterisk server, I need to provide him dialtone
when the calls hits the server. How can I achieve this?

Let's say something like this:

Exten => s,1,Answer
Exten => s,2, "Provide Dial tone"
Exten => s,3, "Dial the number the person will enter after receiving the
dial tone"
Exten => s,4,Hangup

Any ideas?

Thanks very much

Oswaldo


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RE: [Asterisk-Users] Includes include the includes?

[Tarpo, Louie]

If you set up only internal extesions in the default context, then include 
default in [building1] and [office] all of those extensions can call 
internally.  I set up several standard features into the default context which 
everyone can access.

If you want to control feature access, say, for example a line that reads the 
time (let's just say), put that in a different context.

[office]
include => default
include => local
include => international
include => timeline

[building1]
include => default
include => local
 
[default]
exten => 700,1,Dial(SIP/${EXTEN})
exten => 100,1,Dial(SIP/${EXTEN})
exten => 200,2,Dial(SIP/${EXTEN})
;and so on for your other extensions

[timeline]
exten => something

Or, if you wanted to control access to who could call which internal extension, 
then you break out default into groups of their own.

[office]
include => office-ext
include => local
include => international
include => timeline

[building1]
include => building1-ext
include => office-ext
include => local

[office-ext]
exten => 700,1,Dial(SIP/${EXTEN})

[building1-ext]
exten => 100,1,Dial(SIP/${EXTEN})

In that example, office can call other office numbers, make local calls, make 
international calls, and access the timeline feature.  Building 1 can access 
building1 extensions, office extensions, and make local calls.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris Mason
(Lists)
Sent: Friday, June 17, 2005 5:40 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Includes include the includes?


First, let me apologize for the multiple posts - my procmail recipe had a
bug that hid most mail form the list for a day.

The inheritance of includes creates a problem for me. I want to group the
extensions, not put them all in default to control access to features. So
[office] extensions should have the include => longdistance but  [building1]
should not.

However, how can [building1] then dial office?


Chris Mason
www.anguillaguide.com
Tel:  (305) 704-7249 Fax: (815)301-9759  

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Tarpo, Louie
> Sent: Wednesday, June 15, 2005 9:03 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Includes include the includes?
> 
> Yes it does.  You want something like this...
> 
> [office]
> include => default
> include => local
> include => international
> 
> [building1]
> include => default
> include => local
> 
> [default]
> exten => 700,1,Dial(SIP/${EXTEN})
> exten => 100,1,Dial(SIP/${EXTEN})
> exten => 200,2,Dial(SIP/${EXTEN})
> ;and so on for your other extensions
> 

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RE: [Asterisk-Users] Dell PowerEdge + TDM

[Oswaldo Arratia]

I bought a Dell SC1425 and installed a T1/E1 card from Digium and I tried to
configure it using [EMAIL PROTECTED] scripts and did not work, so I went the 
long way and
configure with zaptel's instructions and voila! It works like a charm.

Oswaldo 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Hajek
Sent: Friday, June 17, 2005 8:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Dell PowerEdge + TDM

Hi,

what new Dell servers are compatible and KNOWN to work with Digium TDM
cards? I've looked at Digium's compatibility list at
http://www.digium.com/index.php?menu=compatibility. Does this mean that
other Dell servers like SC1420, SC1425, 800, 1800 are working just fine with
TDM cards?

Can someone clarify this?

Thanks

-David

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[Asterisk-Users] No ringing tone on outgoing SIP trunk

[Normando Marcolongo]

Hi!

I have configured a SIP trunk with a dialing rule.
The trunk behaves normally for incoming calls but when in used for 
outgoing call a strange thing happens.
When I place a call I cannot hear the tone confirming that the remote 
phone is ringing. I simply hear the voice as soon as the party picks up.
When the remote phone start ringing Asterisk receives a SIP packet 
stating that the call is making progress and puts through the incoming 
RTP I suppose.

With an external anonymous ATA from the VOIP provider all works normally.
Here is the debug:

-- Accepted AUTHENTICATED TBD call from 192.168.19.130
-- Accepting DIAL from 192.168.19.130, formats = 0x4
-- Executing Macro("IAX2/[EMAIL PROTECTED]/2";, "dialout-trunk|2|0639006374|") in 
new stack

-- Executing GotoIf("IAX2/[EMAIL PROTECTED]/2";, "1?3:2)") in new stack
-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro("IAX2/[EMAIL PROTECTED]/2";, "record-enable|200|OUT") in new 
stack
-- Executing GotoIf("IAX2/[EMAIL PROTECTED]/2";, "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing GotoIf("IAX2/[EMAIL PROTECTED]/2";, "1?5:8") in new stack
-- Goto (macro-record-enable,s,5)
-- Executing DBget("IAX2/[EMAIL PROTECTED]/2";, "RecEnable=RECORD-OUT/200") in new 
stack

-- DBget: varname=RecEnable, family=RECORD-OUT, key=200
-- DBget: Value not found in database.
-- Executing SetVar("IAX2/[EMAIL PROTECTED]/2";, 
"CALLFILENAME=OUT200-20050616-180439-1118959479.2") in new stack

-- Executing Goto("IAX2/[EMAIL PROTECTED]/2";, "s|14") in new stack
-- Goto (macro-record-enable,s,14)
-- Executing GotoIf("IAX2/[EMAIL PROTECTED]/2";, "0?15:99") in new stack
-- Goto (macro-record-enable,s,99)
-- Executing NoOp("IAX2/[EMAIL PROTECTED]/2";, "NO RECORDING NEEDED") in new 
stack
-- Executing GotoIf("IAX2/[EMAIL PROTECTED]/2";, "1?7") in new stack
-- Goto (macro-dialout-trunk,s,7)
-- Executing GotoIf("IAX2/[EMAIL PROTECTED]/2";, "0?9") in new stack
-- Executing SetCallerID("IAX2/[EMAIL PROTECTED]/2";, "0872596100") in new stack
-- Executing SetGroup("IAX2/[EMAIL PROTECTED]/2";, "OUT_2") in new stack
-- Executing CheckGroup("IAX2/[EMAIL PROTECTED]/2";, "") in new stack
-- Executing SetVar("IAX2/[EMAIL PROTECTED]/2";, "DIAL_NUMBER=0639006374") in new 
stack

-- Executing SetVar("IAX2/[EMAIL PROTECTED]/2";, "DIAL_TRUNK=2") in new stack
-- Executing AGI("IAX2/[EMAIL PROTECTED]/2";, "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf
-- AGI Script fixlocalprefix completed, returning 0
-- Executing SetVar("IAX2/[EMAIL PROTECTED]/2";, "OUTNUM=0639006374") in new 
stack
-- Executing Cut("IAX2/[EMAIL PROTECTED]/2";, "custom=OUT_2|:|1") in new stack
-- Executing GotoIf("IAX2/[EMAIL PROTECTED]/2";, "0?19") in new stack
-- Executing Dial("IAX2/[EMAIL PROTECTED]/2";, "SIP/micso/0639006374") in new 
stack
-- Called micso/0639006374
-- SIP/micso-0626 is making progress passing it to IAX2/[EMAIL PROTECTED]/2
== Spawn extension (macro-dialout-trunk, s, 17) exited non-zero on 
'IAX2/[EMAIL PROTECTED]/2' in macro 'dialout-trunk'
== Spawn extension (from-internal, 90639006374, 1) exited non-zero on 
'IAX2/[EMAIL PROTECTED]/2'

-- Executing Macro("IAX2/[EMAIL PROTECTED]/2";, "hangupcall") in new stack
-- Executing ResetCDR("IAX2/[EMAIL PROTECTED]/2";, "w") in new stack
-- Executing NoCDR("IAX2/[EMAIL PROTECTED]/2";, "") in new stack
-- Executing Wait("IAX2/[EMAIL PROTECTED]/2";, "5") in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on 
'IAX2/[EMAIL PROTECTED]/2' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 
'IAX2/[EMAIL PROTECTED]/2'

-- Hungup 'IAX2/[EMAIL PROTECTED]/2'

What can I do? Do you need somethign else?
Thanks!
Norm
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[Asterisk-Users] auto-dial dial status

[Marco Parmeggiani]

I'm using autodial in conjuction with TxFax to send faxes on demand.
An home made application generates the call file and puts it in the 
outgoing spool, the file is like this:


Channel:Zap/g1/1232314324
MaxRetries:0
RetryTime:60
WaitTime:20
Context:faxout
Extension:s
SetVar:FAX_FILE=/shared/awfax/test.tif

the extension called is this:

[faxout]
exten => s,1,TxFax(/shared/awfax/test.tif|caller)
exten => s,2,Hangup

My problem is that if asterisk can't connect to the called end then it 
doesn't go to the extension, so i am unable to report the error if the 
called end does not respond, does not exist or refuse the call.

Is there some trick (or an elegant solution as well) to solve this problem?

ciao
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Re: [Asterisk-Users] Multiple Sipura 3000

[Tim P]

I saw this in the list awhile back, it helped me setup my sipura 3000s
to act as trunks
Setup the PSTN side of the Sipura 3000 as a trunk within Asterisk
In AMP add an extension (e.g. 200) to correspond to Line 1 on the SPA,
ensure that port is 5060 and context is from-internal. It should be
named SatelliteOut. Disable voicemail and directory.  Add a second
extension (e.g. 201) for PSTN Line on SPA, ensure that port is 5061
and set context to from-pstn. It should be named SatelliteIn.  Disable
voicemail and directory.

In Trunks add a Sip trunk and copy the Outgoing block as follows (just
leave Incoming as it is - do not delete the any defaults, but you do
not need to change them either).:

Trunk name SatelliteOut

context=from-pstn 
fromuser=201 (or whatever extension you used) 
host=IP address of you SPA (needs to be fixed IP) 
port=5061 
secret=your password 
type=peer 
username=201 (or whatever extension you used) 

Inbound User context SatelliteIn 

Leave defaults in Inbound box and leave Register String blank. 

In DID Routes, add DID with a unique string (I used S followed by the
PSTN number that the SPA is attached to - e.g. S888777

Set an outbound route using the new SatelliteOut trunk. 

On the SPA 3000: 
Do the following configuration in admin login, advanced mode: 
In Line 1, make sure SIP port is 5060, & proxy points to your * Box,
NO outbound proxy. Fill out subscriber info with settings above e.g.
User ID = 200
Password =1234
Display Name = SatelliteIn

In PSTN Line, ensure SIP Port = 5061 & proxy = Asterisk Box IP, NO
outbound proxy. Fill out subscriber info with
Display Name = SatelliteOut
User ID = 201
Password =1234
It is vital that you Set Dial Plan 8 to (S0<:S888777>) (for the
string you used for the DID route in Asterisk).

Ensure that both VoIP-To-PSTN Gateway Enable and PSTN-To-VoIP Gateway
Enable are set to yes.
Set PSTN Caller Default DP to 8. 
If you want incoming calls to all be sent to * then set PSTN Ring Thru
Line 1 to no.
Set PSTN Answer Delay to the number of seconds that you want the phone
to ring for before sending it to your * box. Set it to 1.

Leave other settings on the SPA at factory defaults until you really
know what you're doing and want to fine-tune things.

Lastly, make sure you plug into the line jack into the SPA and not the
jack marked phone! I know this seems obvious, but I've missed this
simple step before!

The only kink with inbound using the settings posted is that you can't
have it ring to a phone plugged into the Sipura's phone port. You can
still call out, and the system will still pick up the call if you have
auto attendant recieve the calls. But, if you set the inbound calls to
ring extension 200, your calls will just go directly to voicemail.

That aside, you can have any other phone on the system ring for
inbound calls directly, or set a ring group.


On 6/17/05, Rich Adamson <[EMAIL PROTECTED]> wrote:
> > If I have multiple Sipura 3000 device how can I dial out properly? I
> > can receive call without any problem and that's working really well.
> > Caller ID is shown and when someone call he get's the welcome message
> > the same way I have it configure with the X100P card. I don't seem to
> > have any echo problem with the Sipura 3000 (but I do with X100P cards)
> >
> > My main concern is for outgoing call. Can I create a group like I did
> > in zaptel for Sipura 3000 device? Like if the FXO port of the first
> > Sipura 3000 is busy it will switch to the second and if second is
> > also busy then to the third one, and all the way until all the Sipura
> > 3000 are in used before saying that there's no line left?
> >
> > The only configs I saw on the wiki were with 1 Sipura 3000 but I
> > couldn't find anything on how to setup multiple Sipura 3000 devices
> > in asterisk for outgoing calls.
> 
> If I understood what you're trying to accomplish, try something like
> this.
> 
> In sip.conf, define each spa3k something like this:
> [3021]  ; PSTN side of SPA3000
> type=friend
> host=dynamic
> username=3021
> secret=myspa1
> context=from-sip
> canreinvite=no
> group=17
> pickupgroup=2
> deny=0.0.0.0/0.0.0.0
> permit=216.21.194.0/255.255.255.0
> 
> and be sure to include "group=17" in each spa3k definition.
> 
> Then in extensions.conf, use a dial statement like this:
> 
> exten => _9.,1,Dial(SIP/g17/${EXTEN:1}
> 
> Pick whatever group number you want instead of =17 in the above.
> If I recall correctly, you can have up to 32 groups (or something
> like that).
> 
> When the spa3k first hit the market, someone recommended using port
> 5060 and 5061 in the spa definitions. I have never had to do that
> with any spa3k. Rather, I leave both the fxs and fxo definitions
> in the spa3k default to 5060 and use different userid & secrets
> for the fxs and fxo definitions. The above definition for x3021
> is the actual one in use right now, which functions correctly.
> I've added the "group=17" in the above as an example; I don't
> actua

[Asterisk-Users] Asterisk box as a billing machine in a PSTN network

[Africa Digital]

Hi,
 
Is it possible to use an Asterisk box as billing gateway in a PSTN network? (Asterisk box somewhat connected to PSTN switch)?
 
In case the answer to the above question is yes, how to proceed?
 
Thanks,
 
Simon
		 
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger 
Téléchargez le ici ! 
 
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Re: [Asterisk-Users] Strange problem with G711/G729, Cisco and Grandstream

[Jason Williams]

> But when BT-100 calls 7960 the following is happening:
> 
>-- Executing Dial("SIP/3710-8f2b", "SIP/1707|15") in new stack
>-- Called 1707
>-- SIP/1707-e96a is ringing
>-- SIP/1707-e96a answered SIP/3710-8f2b
>-- Attempting native bridge of SIP/3710-8f2b and SIP/1707-e96a
> 
> May  4 16:46:58 WARNING[5220]: rtp.c:1545 ast_rtp_bridge: codec0 = 256 is
> not codec1 = 4, cannot native bridge.
> 
> sipsrv1*CLI> sip show channels
> 
> Peer User/ANRCall ID  Seq (Tx/Rx)   Format  Last Msg
> 192.168.128.171  170702fff7f7169  00102/0   ulawTx: ACK
> 67.126.23.2513710b5d3f977ea1  00101/52181   g729Rx: ACK
> 
> When this bug is gonna be fixed?
> 

Change the codec order in the phone configuration and place g729
higher it is not asterisk doing this
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Re: [Asterisk-Users] Dell PowerEdge + TDM

[David Hajek]

Thanks for the reply. I'm more interested in lower series then 2850, 
like PE SC1420, PE 800. I don't

need so much power. ;-)

-David


Here we have PowerEdge 2850's doing the donky work with a Wildcard
TE405P in each.

I have seen no operational issues at all with the system or the cards.
We are running CentOS 3 as the operating system and the stable
version of asterisk

The only "niggle" is that when the cards are modprobed on start up
they sometimes <2 in a 100 give an "NMI" message, causing an error
code on the servers little window, its not affected the stability at
all, and its on my list of things to do to find out what causes it!

The systems generally have around 400 - 500 SIP extensions comming off
the back, running around a dual xeon 3Ghz and 3Gb of ram (no
transcoding all G711.u) - we are very happy!

David

On 17/06/05, David Hajek <[EMAIL PROTECTED]> wrote:
 


Hi,

what new Dell servers are compatible and KNOWN to work with Digium TDM
cards? I've looked at Digium's compatibility list
at http://www.digium.com/index.php?menu=compatibility. Does this mean
that other Dell servers like SC1420, SC1425, 800, 1800 are working just
fine with TDM cards?

Can someone clarify this?

Thanks

-David

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[Asterisk-Users] Need Last App-Fax Source

[Damian Minkov]

Can show me some link for the Last FaxApp sources (working with last 
spandsp - i think pre18)
This link is not working. ftp://ftp.soft-switch.org/pub/spandsp. 

The Domain soft-switch.org  is 
not resovable.


And what is the version of libtiff I must install?
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Re: [Asterisk-Users] Dell PowerEdge + TDM

[David John Walsh]

Here we have PowerEdge 2850's doing the donky work with a Wildcard
TE405P in each.

I have seen no operational issues at all with the system or the cards.
 We are running CentOS 3 as the operating system and the stable
version of asterisk

The only "niggle" is that when the cards are modprobed on start up
they sometimes <2 in a 100 give an "NMI" message, causing an error
code on the servers little window, its not affected the stability at
all, and its on my list of things to do to find out what causes it!

The systems generally have around 400 - 500 SIP extensions comming off
the back, running around a dual xeon 3Ghz and 3Gb of ram (no
transcoding all G711.u) - we are very happy!

David

On 17/06/05, David Hajek <[EMAIL PROTECTED]> wrote:
> Hi,
> 
> what new Dell servers are compatible and KNOWN to work with Digium TDM
> cards? I've looked at Digium's compatibility list
> at http://www.digium.com/index.php?menu=compatibility. Does this mean
> that other Dell servers like SC1420, SC1425, 800, 1800 are working just
> fine with TDM cards?
> 
> Can someone clarify this?
> 
> Thanks
> 
> -David
> 
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[Asterisk-Users] RE: Asterisk Google API applications - $4500 bounties available

[Dean Collins]

Btw here is an article on google maps that
I wrote about the other day.

 

http://www.smh.com.au/news/Technology/Map-hacks-make-data-come-alive/2005/06/16/1118869033845.html


 

This is one of the best examples for www.craigslist.com I have ever seen http://housingmaps.com/

 

 

Cheers,

Dean

 

 











From: Dean Collins 
Sent: Wednesday, 1 June 2005 10:11
AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Asterisk Google API
applications - $4500 bounties available



 

In conjunction with my last post on Tellme I want to write
another suggestion for an application I had.

 

I don’t know if you guys have come across Google Gas http://www.ahding.com/cheapgas 

 

But basically it is an application that this guy has
developed using the Google API to search an online database on gas prices in
your area. 

 

One of my strong beliefs about how Asterisk is going to
leave the “Commercial” IP-PBX vendors behind is by leveraging the
open source community to write voice driven applications for Asterisk. The
weather app written for [EMAIL PROTECTED] is great example. (http://sourceforge.net/forum/message.php?msg_id=3004652
the WAF on this was worth setting up asterisk alone, she checks this every
morning for NY weather).

 

I was also hoping that the www.tellme.com
and www.studio.tellme.com tools
would also stimulate this area. People should also check out www.angel.com for other ideas on best of breed
speech applications.

 

The suggestion I would like to make is that someone use the
Google api to write code for a directions application.

 

You could use Tellme to deliver the current address and the
destination address into the Google API and then use text to speech to read
back the directions. With enough finessing this could compete with any of the
current commercial direction solutions that are out there and because it’s
asterisk your cost base could be extremely minimal.

 

Hell you might even get paid for it  http://code.google.com/summerofcode.html

 

 

Just a suggestion, any thoughts? Are there any other speech
driven apps being used today?

 

Cheers,

Dean

 

 








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[Asterisk-Users] Agents/Queues Contexts

[Waldo Rubinstein]

Is there a way to define multiple contexts for agents/queues such  
that in a multi-tenant environment, there could be two different,  
say, Agents 1000?


I'm setting up a multi-tenant configuration and I'm giving each  
tenant a web-based interface to define their own agents and I  
wouldn't want to restrict one tenant from choosing an agent because  
another tenant (which the tenant has no idea about) has already  
chosen that agent id.


Thanks,
Waldo
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Re: [Asterisk-Users] Analog modems behind an Asterisk server?

[Rich Adamson]

> Hello, we would like to hook up analog modems behind an Asterisk server, 
> and we're very interested in the experiences that others have made when 
> attempting that. We assume that there are no inherent problems with 
> modems in respect to the Asterisk software, but it appears that the 
> FXO/FXS hardware restricts this kind of a setup to lower data 
> transmission rates, is this correct?
> 
> Currently, we only transmit at 1200bps, is this rate problematic with 
> Digium cards? Up to what data transmission rate are Digium cards known 
> to work reliable? We do not think we'll ever go beyond 9600bps, can we 
> do this with a let's say TDM400P?
> 
> Will future Digium hardware improve the situation or will this stay the 
> same in the future?
> 
> How is hardware from other vendors performing when using analog modems?

The success rate of moving modem data through a digium analog TDM card 
(fxo & fxs ports) varies and appears to be somewhat related to the
exact motherboard in use. The card very very frequently has an issue
with missed data across the pci bus (card to motherboard). The missed
data negatively impacts any modem call regardless of whether its a
fax machine or pc modem. You are likely to have less then a 50% chance
of making work correctly.

Before a pile of people jump in to say "it works for me", keep in mind
that various types of modems use different modulation schemes and some
are more sensitive than others to distorted audio (missed data). In
very general terms, the higher the modem speed the more likely it will
be negatively impacted by the distortion (missed data).

If you're not familiar with modem technology, I might add there are two
primary items that are directly related to the modem's audio across
analog lines. The "baud" rate of analog signal on the wire and the
bit rate of encoded digital data. You might have a current modem that
allows you to change the bit rate (digital side), but on most modems
you have no control over the analog baud rate (or modulation scheme).
So, changing the modem's bit rate won't impact how well the modem
actually works through the TDM card.

Some people have reported that point of sale and credit card authorization
boxes have worked via the TDM card. However, the modem's used in that
equipment typically are very slow speed modems that were intended to
function in any business environment including those with noisy 
telephone lines. Those have a higher possibility of success, but
should not be interpreted as being the same as a modem used with PC's,
etc.

Bottom line... you will have far less then a 50% chance of making any
PC modem work at acceptable speed through a TDM card.

The latest code for the Sipura boxes (spa3k) appear to have addressed
modem signals (fxs to fxo). I just upgraded two spa3k's to that latest
firmware, but have not attempted to use any modem through it. Might
check to see if anyone else on the list have tried it. The firmware
was just released in the last day or two, so it might take a little
while for folks to try it.


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RE: [Asterisk-Users] Analog modems behind an Asterisk server?

[Florian Overkamp]

Hi, 

> -Original Message-
> Currently, we only transmit at 1200bps, is this rate problematic with 
> Digium cards? Up to what data transmission rate are Digium 
> cards known 
> to work reliable? We do not think we'll ever go beyond 
> 9600bps, can we 
> do this with a let's say TDM400P?

On a pure TDM path this should be fine. In fact I think you should not have
any real limitation if you set everything correctly.

Using VoIP in parts of the link does limit the connection, although we have
seen 14k4 connections run stable for a long time. YMMV.

Florian


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[Asterisk-Users] Analog modems behind an Asterisk server?

[Christian Schnell]

Hello, we would like to hook up analog modems behind an Asterisk server, 
and we're very interested in the experiences that others have made when 
attempting that. We assume that there are no inherent problems with 
modems in respect to the Asterisk software, but it appears that the 
FXO/FXS hardware restricts this kind of a setup to lower data 
transmission rates, is this correct?


Currently, we only transmit at 1200bps, is this rate problematic with 
Digium cards? Up to what data transmission rate are Digium cards known 
to work reliable? We do not think we'll ever go beyond 9600bps, can we 
do this with a let's say TDM400P?


Will future Digium hardware improve the situation or will this stay the 
same in the future?


How is hardware from other vendors performing when using analog modems?

Thanks for any information on this,
Christian Schnell.
REKOBA GmbH Berlin
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[Asterisk-Users] Dell PowerEdge + TDM

[David Hajek]

Hi,

what new Dell servers are compatible and KNOWN to work with Digium TDM 
cards? I've looked at Digium's compatibility list
at http://www.digium.com/index.php?menu=compatibility. Does this mean 
that other Dell servers like SC1420, SC1425, 800, 1800 are working just 
fine with TDM cards?


Can someone clarify this?

Thanks

-David

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Re: [Asterisk-Users] Multiple Sipura 3000

[Rich Adamson]

> If I have multiple Sipura 3000 device how can I dial out properly? I  
> can receive call without any problem and that's working really well.  
> Caller ID is shown and when someone call he get's the welcome message  
> the same way I have it configure with the X100P card. I don't seem to  
> have any echo problem with the Sipura 3000 (but I do with X100P cards)
> 
> My main concern is for outgoing call. Can I create a group like I did  
> in zaptel for Sipura 3000 device? Like if the FXO port of the first  
> Sipura 3000 is busy it will switch to the second and if second is  
> also busy then to the third one, and all the way until all the Sipura  
> 3000 are in used before saying that there's no line left?
> 
> The only configs I saw on the wiki were with 1 Sipura 3000 but I  
> couldn't find anything on how to setup multiple Sipura 3000 devices  
> in asterisk for outgoing calls.

If I understood what you're trying to accomplish, try something like
this.

In sip.conf, define each spa3k something like this:
[3021]  ; PSTN side of SPA3000
type=friend
host=dynamic
username=3021
secret=myspa1
context=from-sip
canreinvite=no
group=17
pickupgroup=2
deny=0.0.0.0/0.0.0.0
permit=216.21.194.0/255.255.255.0

and be sure to include "group=17" in each spa3k definition.

Then in extensions.conf, use a dial statement like this:

exten => _9.,1,Dial(SIP/g17/${EXTEN:1}

Pick whatever group number you want instead of =17 in the above.
If I recall correctly, you can have up to 32 groups (or something
like that).

When the spa3k first hit the market, someone recommended using port
5060 and 5061 in the spa definitions. I have never had to do that
with any spa3k. Rather, I leave both the fxs and fxo definitions
in the spa3k default to 5060 and use different userid & secrets
for the fxs and fxo definitions. The above definition for x3021
is the actual one in use right now, which functions correctly.
I've added the "group=17" in the above as an example; I don't
actually use that right now (for different reasons).


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[Asterisk-Users] Call group channel limits

[Iain Sims]

I have a question that I've so far been unable to find the answer to:

Using an E1 interface for my PSTN connection I want to setup 5 SIP
phones in a call group (with a unique number for inbound calls) but only
allow the call group to receive a maximum of 3 calls at any one time.

Does Asterisk have this functionality and if so, could someone point me
to some examples??

Regs.

Iain.
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RE: [Asterisk-Users] Includes include the includes?

[Chris Mason (Lists)]

First, let me apologize for the multiple posts - my procmail recipe had a
bug that hid most mail form the list for a day.

The inheritance of includes creates a problem for me. I want to group the
extensions, not put them all in default to control access to features. So
[office] extensions should have the include => longdistance but  [building1]
should not.

However, how can [building1] then dial office?


Chris Mason
www.anguillaguide.com
Tel:  (305) 704-7249 Fax: (815)301-9759  

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Tarpo, Louie
> Sent: Wednesday, June 15, 2005 9:03 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Includes include the includes?
> 
> Yes it does.  You want something like this...
> 
> [office]
> include => default
> include => local
> include => international
> 
> [building1]
> include => default
> include => local
> 
> [default]
> exten => 700,1,Dial(SIP/${EXTEN})
> exten => 100,1,Dial(SIP/${EXTEN})
> exten => 200,2,Dial(SIP/${EXTEN})
> ;and so on for your other extensions
> 

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Re: [Asterisk-Users] Dial Commands "D" Option Question

[Rich Adamson]

> When using the dial command and the D option to send DTMF digits when
> the channel is answered, is there a way to allow for some dead air,
> and then send more DTMF digits? I would like to automate a call, and
> it requires entry of a few short dtmf digits all a couple seconds
> apart from each other.

Might look at the "w" within the dial string, which adds delay for each
"w" appearing in the string. Something like:
 exten => _1NX,3,Dial(IAX2/xyz-itsp/9ww${EXTEN})


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[Asterisk-Users] Miax: Digital voice channel when connecting to asterisk

[Roger Schreiter]

Hi,

I've bought a Siemens GSM-modem based on the Siemens TC35-module.

I studied the operation manual of the modem and found, that for
transferring voice via the RS232 wire, the module supports
"RS232-mulitplexing" and wires the voice data on a separate
"channel" (whatever this means on RS232?).

Now I wonder, whether that feature is supported by miax.
All what I read about, was transfering GSM voice data via bluetoth
from a cell phone to miax.

Does anyone succeeded in connecting a GSM-modem via miax to
asterisk and transfering the GSM voice data via the RS232
cable?


Thanks for any hints!
Roger.


P.S.
Somewhere I read the advice, that I should connect the (analogue) audio
connector to the PC's soundcard, which is supported by miax. But
transferring the GSM voice data analogious and digitizing again
afterwords is not, what I really am looking for.

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Re: [Asterisk-Users] Bill seconds

[David John Walsh]

Americo

60+60 isn't a VoIP term directly but a generic one within the telephony industry

if it were 60+30 it would mean the following

You are billed for 60 second as soon as the call is answered, even if
you only stay on the line for 7 seconds

The +30 then referes to the onward billing cycle, so in this case you
are billed in blocks of 30 seconds (ie if you call is 1 min 15 seconds
you are billed for 1 min 30)

You said that you are billed for a whole second minuite if you go over
by even 1 second, so that would be a +60, and since its always a
bigger or equal number first we are guessing that you are in 60+60
rate plan

I think its more common in your part of the world for your carriers to
bill 30+6.  One in the replys suggested a very favorable rate of 6+6

The important thing to rember here is that you can't gaurentee enough
return if you do a billing rate that is better than that of your
carriers - it sounds to me like your offering your service on a 1+1
(ie true per second billing) rate - very honarable, but your carrier
needs to offer the same.

I hope that helps

On 17/06/05, Americo Sanchez C. <[EMAIL PROTECTED]> wrote:
> 
> 
> >From: "Leon Sun" <[EMAIL PROTECTED]>
> >Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> >
> >To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> >
> >Subject: RE: [Asterisk-Users] Bill seconds
> >Date: Thu, 16 Jun 2005 10:56:23 -0700
> >
> >The easiest way is to change another vendor asap.
> Do you mean to change to another telecom? In my country there is a telephone
> monopoly :( Telefonica del Peru)
> >It is ridiculous that your
> >carrier still uses 60+60 now(30+6 is an asset). 2 seconds doesn't matter
> >and
> >billing unit does.
> Sorry I am not an expert in VoIP, What is the meaning of "60+60"?
> >
> >
> >Leon Sun
> >
> >-Original Message-
> >From: [EMAIL PROTECTED]
> >[mailto:[EMAIL PROTECTED] On Behalf Of Darren Wiebe
> >Sent: June 15, 2005 10:06 PM
> >To: Asterisk Users Mailing List - Non-Commercial Discussion
> >Subject: Re: [Asterisk-Users] Bill seconds
> >
> >I've done a little thinking on this one  If you are using ASTCC, it
> >would be fairly straightforward to edit it and have it make a 2 second
> >adjustment.  If your using another solution it probably would be fairly
> >easy also...
> >
> >Darren Wiebe
> >[EMAIL PROTECTED]
> >
> >Americo Sanchez C. wrote:
> >
> > >
> > > Hi all,
> > >
> > > We've installed Asterisk on a rural development project and we're
> > > testing a prepaid phone service. As far as now we're having terrific
> > > service results but there's a problem with the calls billing at our
> > > local telecom. For instance, a farmer buys a 1 dollar phone card and use
> > > it to dial a USA number, the call should lasts for 60 seconds. Asterisk
> > > is doing a great job finishing the call exactly at 60 seconds. The
> > > problem is that the telecom company billing system adds a two second
> > > delay for each call, so the bill is not for 1 but 2 minutes (they round
> > > fractions up).
> > >
> > > We're loosing money and the local telecom doesn't seem to have a
> > > solution for this matter.
> > >
> > > Have you experienced something similar? Do you have any idea of how can
> > > we solve this? Is it possible to configure Asterisk so that the system
> > > thinks that a minute has 58 seconds instead of 60?
> > >
> > > _
> > > MSN Amor: busca tu ½ naranja http://latam.msn.com/amor/
> > >
> > > ___
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> > >
> >
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> 
> _
> Consigue aquí las mejores y mas recientes ofertas de trabajo en América
> Latina y USA: http://latam.msn.com/empleos/
> 
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[Asterisk-Users] bristuff-0.2.0-RC8g: zaptel error in suse 9.2

[Manuel Casal]

Hi, I have the SuSe9.2 installed in a box with a QuadBri.
I have followed all the instructions i have found and this is my best 
result... only one error compiling zaptel :( .. y have the kernel 
sources an already made the links to its


drwxr-xr-x   8 root root 328 Jun 16 20:20 .
drwxr-xr-x  12 root root 320 Jun 16 14:05 ..
drwxr-xr-x   3 root root 136 Jun 16 20:17 asterisk
drwxr-xr-x  22 root root 664 Jun 16 20:20 kernel-modules
lrwxrwxrwx   1 root root  14 Jun 16 18:01 linux -> linux-2.6.8-24
lrwxrwxrwx   1 root root  14 Jun 16 18:03 linux-2.6 -> linux-2.6.8-24
drwxr-xr-x  21 root root 864 Jun 17 10:27 linux-2.6.8-24
drwxr-xr-x   3 root root  72 Oct  6  2004 linux-2.6.8-24-obj
drwxr-xr-x   3 root root  72 Jun 16 17:28 linux-2.6.8-24.16-obj
lrwxrwxrwx   1 root root  18 Jun 16 18:01 linux-obj -> linux-2.6.8-24-obj
drwxr-xr-x   7 root root 168 Jun 16 14:08 packages

I made the "make menuconfig" and "make dep" in the kernel sources.

but when i try to complile zaptel  from the britstuff package  this is 
the result:


linux:/usr/src/asterisk/bristuff-0.2.0-RC8g/zaptel-1.0.7 # make clean
rm -f torisatool makefw tor2fw.h
rm -f zttool ztspeed zttest ztmonitor
rm -f *.o ztcfg tzdriver sethdlc sethdlc-new
rm -f zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo
rm -f *.ko *.mod.c .*o.cmd
rm -f gendigits tones.h
rm -f libtonezone*
rm -f tor2ee
rm -f core
rm -rf .tmp_versions

linux:/usr/src/asterisk/bristuff-0.2.0-RC8g/zaptel-1.0.7 # make
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o gendigits.o gendigits.c

cc -o gendigits gendigits.o -lm
./gendigits
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"makefw.c   -o makefw

./makefw tormenta2.rbt tor2fw > tor2fw.h
Loaded 69900 bytes from file
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o ztcfg.o ztcfg.c
cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DBUILDING_TONEZONE -o zonedata.lo 
zonedata.c
cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DBUILDING_TONEZONE -o tonezone.lo 
tonezone.c

ar rcs libtonezone.a zonedata.lo tonezone.lo
cc -o ztcfg ztcfg.o -lm -L. libtonezone.a
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o torisatool.o torisatool.c

cc -o torisatool torisatool.o
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o ztmonitor.o ztmonitor.c

cc -o ztmonitor ztmonitor.o
cc -c ztspeed.c
cc -o ztspeed ztspeed.o
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o zttool.o zttool.c

cc -o zttool zttool.o -lnewt
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"zttest.c   -o zttest
You do not appear to have the kernel sources for your current kernel 
installed.

make: *** [linux26] Error 1
linux:/usr/src/asterisk/bristuff-0.2.0-RC8g/zaptel-1.0.7 #

Why dont found the kernel sources?

Help please

Thanks


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RE: [Asterisk-Users] Dial timeout when server down

[Kris Boutilier]

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Yves
> Sent: Friday, June 17, 2005 2:09 AM
> To: Asterisk - Users
> Subject: [Asterisk-Users] Dial timeout when server down
> 
> 
> Hello,
> 
> When dialing somewhere and the other side is down, Asterisk 
> waits until dial timeout before sending "CHANUNAVAIL". I think that if 
> after several seconds there are not any reply (I mean at the IP level) we 
> could 
> consider that the link is just down and handle the situation.
> 
> Is it possible to configure Asterisk to have this behaviour?
> 

If you're refering to an IAX channel then, yes, that's the concept behind the 
'qualify=' option. However, there are known weaknesses including that the loss 
of a single ping makes the remote host appear to be down. There is a patch for 
head available at http://bugs.digium.com/view.php?id=4192 that attempts to make 
peer qualification more useful.

Hope that helps.

Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
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Re: [Asterisk-Users] Comprehensive Asterisk Load Testing

[Zoa]

The call generator is very not user friendly now and undocumented, i
recommend not to use it and use some simple script somewhere for now.
(you could find some scripts somewhere on astertest.com).
I will put fixing that callgenerator on the (big) todo list.

Zoa,

Matt wrote:


Has anyone gotten this tester to work?  i can get it to log in and
show me my call load.. but it doesn't seem to MAKE any calls.

On 3/30/05, Kristian Kielhofner <[EMAIL PROTECTED]> wrote:



Bicom Systems wrote:



[EMAIL PROTECTED] wrote:




Its a very very bad idea to do this on production boxes. Especially if
you are trying to see how far you can go, and then you cross that tiny
border :)

Your production calls will not like an idle cpu% of 0% and a load of
500.



I could not agree more with you hence my question :)

However, the tests results produced on test boxes:
How realistic it is?
Does it really presents "real life"
scenarios and results?
Does it take in consideration different
type of services (calls, IVR, queues) ?

I am not trying to put down anyone or anything here, I  am just
curious.

Ta
Senad



Senad,

   I have yet to take a real hard look or contact Zoa, but if all you are
doing is calling an extension (very rapidly and many, many times) it
really would not be very hard to test queues, music on hold, meetme,
etc.  I am downloading the callgenerator from astertest.com right now...

   The most realistic test is to (obviously) register as many phones as
possible and hire hundreds of people to talk on them... :)

--
Kristian Kielhofner
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Re: [Asterisk-Users] EuroISDN Italy - quadbri - zaptel.conf - whatsettings work ?

[Emanuele Pucciarelli]

Robert Rozman wrote:
> Is framing and coding (ami,ccs) right for Italy ?

They are dummy settings with bristuff.  The example config will surely do :)

-- 
Emanuele
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[Asterisk-Users] Dial timeout when server down

[Yves]

Hello,

When dialing somewhere and the other side is down, Asterisk waits until 
dial timeout before sending "CHANUNAVAIL". I think that if after several 
seconds there are not any reply (I mean at the IP level) we could 
consider that the link is just down and handle the situation.


Is it possible to configure Asterisk to have this behaviour?

Many thanks.

Yves.
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Re: [Asterisk-Users] Multiple Sipura 3000

[Chris Stenton]

[outgoing]

ignorepat => 9
exten => _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061)
exten => _9.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061)
exten => _9.,3,Playback(nomoreline)
exten => _9.,4,Hangup


Chris


- Original Message - 
From: "Martin Roy" <[EMAIL PROTECTED]>

To: 
Sent: Thursday, June 16, 2005 11:41 PM
Subject: [Asterisk-Users] Multiple Sipura 3000


If I have multiple Sipura 3000 device how can I dial out properly? I  
can receive call without any problem and that's working really well.  
Caller ID is shown and when someone call he get's the welcome message  
the same way I have it configure with the X100P card. I don't seem to  
have any echo problem with the Sipura 3000 (but I do with X100P cards)


My main concern is for outgoing call. Can I create a group like I did  
in zaptel for Sipura 3000 device? Like if the FXO port of the first  
Sipura 3000 is busy it will switch to the second and if second is  
also busy then to the third one, and all the way until all the Sipura  
3000 are in used before saying that there's no line left?


The only configs I saw on the wiki were with 1 Sipura 3000 but I  
couldn't find anything on how to setup multiple Sipura 3000 devices  
in asterisk for outgoing calls.


I would set it up the same way I have currently the zap channel  
configure so like this :


[outgoing]

ignorepat => 9
exten => _9.,1,Dial(Zap/g2/${EXTEN:1})
exten => _9.,2,Playback(nomoreline)
exten => _9.,3,Hangup


I tried this and it's working :

[outgoing]

ignorepat => 9
exten => _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061)
exten => _9.,2,Playback(nomoreline)
exten => _9.,3,Hangup


10.0.1.111:5061 is the IP and SIP port of the Sipura 3000 device. So  
that would work great if I had only one Sipura but if I have multiple  
I would do it that way ? :


[outgoing]

ignorepat => 9
exten => _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061)
exten => _9.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061)
exten => _9.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061)
exten => _9.,4,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061)
exten => _9.,5,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061)
exten => _9.,6,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061)
exten => _9.,7,Playback(nomoreline)
exten => _9.,8,Hangup


would that work? it's not quite the best thing to do as if I leave  
all the Sipura 3000 devices on DHCP if the IP ever change it will  
stop working and if one line is busy what will happen...


Thanks

Martin

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[Asterisk-Users] ata186 IVR problem

[Betül Gözlükoğlu]

Hi;

Connected ata186 sip version 3.1.1 to IVR system...able to
call / receive calls from ata but does not accept any dtmf…

When internal extension is dialed, it is not recognized and
continues IVR music…Does antbody has any idea how to make dtmf
configuration from ata186 sip version?...it was used with sccp and cm without
problem of dtmf…

 

Thanks in advance

Betul

 





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Re: [Asterisk-Users] Viva Madrid!

[Wojciech Tryc]

Agreed,
I will post some pics early next week:)
Wojtek
- Original Message - 
From: "Nicolás Gudiño" <[EMAIL PROTECTED]>

To: 
Sent: Thursday, June 16, 2005 8:27 PM
Subject: [Asterisk-Users] Viva Madrid!


enough said
--
Nicolás Gudiño
Buenos Aires - Argentina
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RE: [Asterisk-Users] Long time to detect hang-up

[Stojan Sljivic - GDS]

Title: Message



Hi,
 
Has 
anyone experienced the same problem.
My 
telco provider is SBC.
 
Regards,Stojan 
Sljivic 

  
  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Stojan 
  Sljivic - GDSSent: Tuesday, June 14, 2005 14:48To: 
  'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: 
  [Asterisk-Users] Long time to detect hang-up
  Hi,
   
  I 
  use Asterisk 1.0.5 and TDM04B.
  When 
  an incoming call over ZAP channel hangs-up, it takes 10 seconds until Asterisk 
  realize that.
   
  How 
  can I shorten the time of hang-up detection?
   
  Regards,
  Stojan Sljivic
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[Asterisk-Users] Junk at the beginning of frame

[Asterisk]

We upgraded to cvs-head a couple of days ago, and now get a whole slew 
of warnings in the error log:



interface.c: Junk at the beginning of frame xx

has anyone else seen this, or do I need to incur the wrath of developers 
and post this to the -dev list ;)


Julian
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[Asterisk-Users] Re: meetme - conf-invalid

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
qrss <[EMAIL PROTECTED]> wrote:
> Yes, meetme requires a clock source.  You could try ztdummy.  I tried
> using an FXO card as a clock source and observed that SIP calls connected
> to the conference seemed to get out of sync.  Basically, after perhaps 20
> minutes or so in conference there was a 2 - 3 second delay between the
> time that one party spoke and the other party heard what was said.  I have
> not tried ztdummy myself.  Has anybody else seen this?

Yes. Try the patch at http://bugs.digium.com/view.php?id=4252 to see
whether it helps. Please post your results to that bug - thanks!

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] Re: meetme - conf-invalid

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Kevin Bockman <[EMAIL PROTECTED]> wrote:
> > Yes, meetme requires a clock source.  You could try ztdummy.  I tried
> > using an FXO card as a clock source and observed that SIP calls connected
> > to the conference seemed to get out of sync.  Basically, after perhaps 20
> > minutes or so in conference there was a 2 - 3 second delay between the
> > time that one party spoke and the other party heard what was said.  I have
> > not tried ztdummy myself.  Has anybody else seen this?
> 
> According to
> http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer%20ztdummy
> , ztrtc is supposted to fix the MeetMe delay.  It compiles by default on
> Linux 2.6 with newer versions of -HEAD.

Only if you remove the #if 0 from around #define USE_RTC

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] Sip INFO DTMF over satellite

[[EMAIL PROTECTED]]

Hi all

Am experiencing a very weird behaviour with sip info DTMF . I have an * 
box which is a satellite hop away. When I make a call from a Grandstream 
call set to send DTMF thru sip info, am able to navigate through the * 
menus very well. meaning that the DTMF is being recieved very well. The 
problem is when I connect to an external sip server that has an IVR. The 
external sip server does not seem to recieve the DTMF from the GS phone.
From an * server on the LAN connecting to the external sip server, the 
DTMF are recieved very well on the external server. I read something 
about the 250ms  of the Sip INfo Mode of DTMF relay but its very shallow 
and have no idea on how to change the settings on Asterisk.


http://www.voip-info.org/tiki-index.php?page=SIP%20Info%20DTMF

Note: dtmf-relay or dtmf are not yet IANA registered application mime types

Cisco uses SIP INFO for DTMF relay: See 
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftinfo.htm 


Cisco equipment uses the following feature restrictions:
-Minimum signal duration is 100 milliseconds (ms). If a request is 
received with a duration less than 100ms, the minimum duration of 100 ms 
is used by default.
-Maximum signal duration is 5000 ms. If a request is received with a 
duration longer than 5000 ms, the maximum duration of 5000 ms is used by 
default.
-If no duration parameter is included in a request, the gateway defaults 
to a signal duration of 250 ms.


Can this be what is causing my problem ?


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RE: [Asterisk-Users] Caller ID

[Stojan Sljivic - GDS]

Hi,

My telco provider is SBC. I think that they use FSK to transmit caller ID.
How can I set-up Asterisk so that I can see caller ID on incoming calls.

Regards,
Stojan Sljivic 


> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Calin Serbanescu
> Sent: Wednesday, June 15, 2005 19:25
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Caller ID
> 
> 
> What standard does your telco send the caller-id in ? ETSI 
> FSK, bellcore... ? 
> 
> On Wed, 2005-06-15 at 16:26 +0200, Stojan Sljivic - GDS wrote:
> > Hi Juan,
> > 
> > I have Caller Id service enabled. When I connect the line 
> to the phone 
> > I see the caller Id on the phone's display.
> > 
> > I have callerid=asreceived. I have also played with various 
> > combinations of cidsignalling and cidstart, but with no success.
> > 
> > Regards,
> > Stojan Sljivic
> > 
> > 
> > > -Original Message-
> > > From: [EMAIL PROTECTED]
> > > [mailto:[EMAIL PROTECTED] On Behalf Of 
> > > Juan Manuel Coronado Z.
> > > Sent: Wednesday, June 15, 2005 15:37
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: Re: [Asterisk-Users] Caller ID
> > > 
> > > 
> > > Hi,
> > > 
> > > First, you must ensure that the your Telco is sending you the
> > > caller ID with incoming calls. In some countries this is an 
> > > aditional service you have to pay for, upon request.
> > > 
> > > If you already have the service from the Telco, check in your
> > > zapata.conf that you have "callerid=asreceived" on your 
> > > channels and group definitions.
> > > 
> > > Hope this will help.
> > > 
> > > Regards,
> > > 
> > > 
> > > Juan Manuel Coronado Z.
> > > 
> > > 
> > > On mar, 2005-06-14 at 14:50 +0200, Stojan Sljivic - GDS wrote:
> > > > Hi,
> > > >  
> > > > I'm using TDM04B and Asterisk 1.0.5.
> > > >  
> > > > How can I setup the Asterisk so that I get caller ID?
> > > > I do not get caller ID currently.
> > > >  
> > > > Regards,
> > > > Stojan Sljivic ___
> > > > Asterisk-Users mailing list Asterisk-Users@lists.digium.com
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
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> > 
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