[Asterisk-Users] Re: Cisco dtmf

[Tomislav Parcina]

In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] 
says...
> I use:
> 
> # Enable_VAD (1-enabled, 0-disabled)
> enable_vad: "0"
> dtmf_inband: "1"
> dtmf_outofband: "never"
> dtmf_avt_payload: "101"
> 
> and it works well for me.  Sometimes going through a callmanager I have
> to set outofband to avt to get dialtone sent though.

Thank you! This one works for me.


-- 

Tomislav Parcina
[EMAIL PROTECTED]

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Re: [Asterisk-Users] Stay away from Grandstream!

[Brian Capouch]

calvis wrote:
Are you saying that we just wasted our money with our recent purchase of 
Grandstream phones?   The last thing I need is problems with a phone.   
Someone please confirm…are these phones unusable?




They're not unusable.

Grandstreams are a budget-line phone, as is obvious from their price.

They don't perform as well as the expensive Ciscos and Polycoms, but 
many of us are using them in a variety of circumstances quite happily.


"YMMV," as they say.

It's always a good idea before adopting *any* phone to get a couple of 
them into a lab environment and play with them first.  Surprises when a 
system is out in the field and under load can be very bad for one's 
peace of mind.


B.
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RE: [Asterisk-Users] Stay away from Grandstream!

[calvis]

Are you saying that we just wasted our
money with our recent purchase of Grandstream phones?   The last thing I need
is problems with a phone.   Someone please confirm…are these phones
unusable?

 

 

 

 

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erick Baum
Sent: Tuesday, December 27, 2005
6:53 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Stay
away from Grandstream!



 



We have 50 Grandstream's that we purchased about 3 months ago. 
They're all at one site and we've had nothing but trouble with all the
phones.  The echo is the worst problem of all.  We had to upgrade
from the default 1.0.1.9 firmware to the 1.0.1.12 "beta" to get the
speakerphone to work properly.  Once we did that, there is now a bad
echo if one of the GXP users turns their volume up too high, the other party
can hear an echo.  If the GXP user turns their volume down a
bit, the echo starts to go away.  This happens internally from GXP to GXP
as well as with outside callers.  At first Grandstream didn't want to
admit that they knew about this or that they'd even ever heard of this problem
before.  But they finally at least said they would have an engineer check
into it.  Then after some research online I come to find that lots of
people are having the echo issue with the newer firmware and they had also
notified Grandstream months ago.  So they know about the problem. 





 





On top of the echo issue, which is completely unacceptable, we've had a
couple phones flat out die, we've had several that had the PoE go dead so we
were forced to use the AC adapter.  And many of the other phones suffer
from all kinds of stupid little intermittent issues such as dropped
calls, reboots and strange ticking and static on the line, even on
internal calls.  We discovered that quite a few of the network cables that
came with the phones seemed to be faulty, which when replacing them seemed to
solve some of our dropped calls and spontaneous reboot problems. 
Some of the phones had bad handset cables.  Replacing some of those
seemed to get rid of some of the static issues.  We've replaced several of
the really troublesome phones with Cisco's or Polycom's, and what do you know,
no problems whatsoever. 





 





All in all, we have been extremely disappointed in the reliability of
the phones and especially unhappy with the level of service from Grandstream. 
Grandstream should not be permitted to sell these phones to consumers, marketed
as a business class phone of all things!  They're knowingly selling a
faulty product.  It's outrageous.  And what's even more amazing
is they claim that the GXP2000 won some sort of award, and it's even a
"Best Seller" at Atacomm.com. 
How it the world can that be.  The phone is a piece of crap.  They're
either going to fix these problems once and for all or give us our money back.





 





Here's a good page regarding many of these issues... including the
echo.. argh... 





http://www.voip-info.org/wiki/view/GXP-2000





 





Erick

 





On 12/27/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: 

>
> What issues are you having with attended call transfer? In recent months
> I've gone through a fair number of GXP2000 firmware versions and I can't
say 
> any of them have had a problem with attended transfer.

I saw this topic and I myself the recommend the same!! Stay away from
Grandstream... unless you are younger, much hair in the head and no
white hair :-) 


Isamar

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[Asterisk-Users] Re: 4-port external sip fxo which doesnt suck?

[Bob Knight]

For a box that has very poor reviews, it sure is great
to use a box that you can throw in the closet and just
forget about it.  They just always work and sound great.
The first time you configure one is a bit of a pain, but
after that it is cruz time.

I use a linux mib browser (mbrowse) because I work in
an usoft free environment.  I can drop ship a unit and
have them plug it into the pbx lan and then configure it
remotely.  I find snmp more convenient than a browser interface.

I have deployed quite a few Mediatrix 1204 and have never
gone back and looked at any of them again.  They just work.


I'm looking for a 4-port external sip fxo which doesn't suck.



o) Clipcomm CG-410. Poor reviews.
o) Mediatrix 1204. Very poor reviews.
o) Audiocodes MP104. Poor reviews.
o) DLink DVG-3004S. Doesnt seem to exist yet.



--
Bob Knight
[-w] the work option
[EMAIL PROTECTED]
925-449-9163

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Re: [Asterisk-Users] Asterisk on VPS

[Dualcall.com]

hello,
yes, I've done Asterisk on VPS (Xen)

Cheers,
Madhawa
Ross C wrote:


Hi all,

I’m curious if anyone has tried installing Asterisk on a Virtual 
Private Server from a web hosting company? I am a web hosting reseller 
with Jodohost.com, so I can have as many Linux VPS’s as I want, and I 
thought I might try it. I’m just curious if anyone else has tried this 
before?


It, obviously, wouldn’t have any PSTN hardware cards or anything; I’d 
just use a VOIP provider like Teliax or Broadvoice. With a VPS 
account, I have complete root access, so I can’t see why it wouldn’t 
work. It would only have 3 or 4 users connected to it, so I don’t 
*think* performance would be an issue. Ping times from the VPS server 
to the VOIP provider’s server (Teliax) are about 45ms, so I think 
everything is OK in that regard as well. Any feedback/opinions?? Thanks!


-Ross



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[Asterisk-Users] Maximizing audio quality

[Wolfgang Borgon]

I'm trying to understand how to maximize the quality of audio  delievered from my asterisk server through a 3rd-party PSTN termination  to a mobile phone.As far as I know the codec, network quality, and original sound file are the main factors -- are there others?In regards to the sound file, are particular formats preferred -- i.e.  RAW, GSM, WAV, MP3?  Based on what I've seen so far, the sound in  the GSM files provided with Asterisk are a little soft, but at least  smooth.  A RAW file I created after converting from MP3 and WAV,  sounded raspy.   Does anyone have any tips for creating the  best quality voice recordings?If my questions are answered elsewhere, please point me in that direction.  Thank you!  
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Re: [Asterisk-Users] 4-port external sip fxo which doesnt suck?

[Chris Mason (Lists)]

[EMAIL PROTECTED] wrote:

I'm looking for a 4-port external sip fxo which doesn't suck.

o) Clipcomm CG-410. Poor reviews.
o) Mediatrix 1204. Very poor reviews.
o) Audiocodes MP104. Poor reviews.
o) DLink DVG-3004S. Doesnt seem to exist yet.

Is anyone actually using a 4 port external sip fxo which doesn't suck?

It almost seems better to buy a pile of SPA-3000 and use them for just 
SIP FXO.


There's a couple from Taiwan that also are not perfect, the VG-400 is 
one Ihave here and I have a couple of Yoda units. Although they work, 
chances are you won't get callerid, the indications will not be right 
for the US, and they work out just as expensive as four SPA-3000s, which 
are very good. Don't hold your breath for firmware updates.
With 4 X SPA3000, if one dies, you can replace just that one, and a 
spare is a lot cheaper that a spare four port.


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


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[Asterisk-Users] Trial Edition of Druid Asterisk Web-interface

[Vikram Rangnekar]

Hi Guys,

We got a lot of feedback asking for free trail editions so here it is. A full
featured trial. We are definatly commited to this product and are focused on
creating the best web/non-web interfaces into asterisk for easy management of
remote and local Asterisk servers.

Your support and feedback is very valuable to us so please send us your ideas
and thoughts.

[EMAIL PROTECTED]

An online demo is also on its way.

http://www.voiceroute.net/
http://www.voiceroute.net/site/trial.php


-- 
regards
Vikram 
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Re: [Asterisk-Users] Stay away from Grandstream!

[Steve Underwood]

Erick Baum wrote:

We have 50 Grandstream's that we purchased about 3 months ago.  
They're all at one site and we've had nothing but trouble with all the 
phones.  The echo is the worst problem of all.  We had to upgrade from 
the default 1.0.1.9  firmware to the 1.0.1.12 
 "beta" to get the speakerphone to work properly.  
Once we did that, there is now a bad echo if one of the GXP users 
turns their volume up too high, the other party can hear an echo.  If 
the GXP user turns their volume down a bit, the echo starts to go 
away.  This happens internally from GXP to GXP as well as with outside 
callers.  At first Grandstream didn't want to admit that they knew 
about this or that they'd even ever heard of this problem before.  But 
they finally at least said they would have an engineer check into it.  
Then after some research online I come to find that lots of people are 
having the echo issue with the newer firmware and they had also 
notified Grandstream months ago.  So they know about the problem.
 


[...]

Well you could buy Polycoms, except they have the same problem.
Or the Snoms... oh, yeah, they have this too
Or the Sipuras... damn, same problem

It would be interesting to see someone compile a list of phone which 
echo cancel the handset properly.


Regards,
Steve


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[Asterisk-Users] 4-port external sip fxo which doesnt suck?

[asterisk]

I'm looking for a 4-port external sip fxo which doesn't suck.

o) Clipcomm CG-410. Poor reviews.
o) Mediatrix 1204. Very poor reviews.
o) Audiocodes MP104. Poor reviews.
o) DLink DVG-3004S. Doesnt seem to exist yet.

Is anyone actually using a 4 port external sip fxo which doesn't suck?

It almost seems better to buy a pile of SPA-3000 and use them for just SIP FXO.

-Dan
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[Asterisk-Users] Asterisk seg fault (SVN-branch-1.2-r7641)

[Robert La Ferla]

   -- Executing VoiceMail("SIP/999-e59b", "500|g4") in new stack

Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread -1212703824 (LWP 4440)]
0x003d0110 in rawmemchr () from /lib/libc.so.6
(gdb) bt
#0  0x003d0110 in rawmemchr () from /lib/libc.so.6
#1  0x003c582b in _IO_str_init_static_internal () from /lib/libc.so.6
#2  0x003bb657 in vsscanf () from /lib/libc.so.6
#3  0x003b6bf2 in sscanf () from /lib/libc.so.6
#4  0x0055e87d in vm_exec (chan=0x8b92fc8, data=0xb7b76fe8) at 
app_voicemail.c:5501
#5  0x0808d033 in pbx_extension_helper (c=0x8b92fc8, con=Variable "con" 
is not available.

) at pbx.c:544
#6  0x0808e4d4 in __ast_pbx_run (c=0x8b92fc8) at pbx.c:2220
#7  0x0808f0dc in pbx_thread (data=0x8b92fc8) at pbx.c:2507
#8  0x004d6b80 in start_thread () from /lib/libpthread.so.0
#9  0x0042e9ce in clone () from /lib/libc.so.6
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Re: [Asterisk-Users] Stay away from Grandstream!

[Elene Kinsky]

Yes, our mistake that we make order directly from GS. Should make via 
some reseller instead. About rate: 2/11 phones dead.  Regarding the 
phone quality, it looks a bit cheap (I know it's really cheap), some 
buttons are worn out after few months of intensive use.

Now maybe we'll consider to buy other phones for our customers.

Erick Baum wrote:

We have 50 Grandstream's that we purchased about 3 months ago.  
They're all at one site and we've had nothing but trouble with all the 
phones.  The echo is the worst problem of all.  We had to upgrade from 
the default 1.0.1.9  firmware to the 1.0.1.12 
 "beta" to get the speakerphone to work properly.  
Once we did that, there is now a bad echo if one of the GXP users 
turns their volume up too high, the other party can hear an echo.  If 
the GXP user turns their volume down a bit, the echo starts to go 
away.  This happens internally from GXP to GXP as well as with outside 
callers.  At first Grandstream didn't want to admit that they knew 
about this or that they'd even ever heard of this problem before.  But 
they finally at least said they would have an engineer check into it.  
Then after some research online I come to find that lots of people are 
having the echo issue with the newer firmware and they had also 
notified Grandstream months ago.  So they know about the problem.
 
On top of the echo issue, which is completely unacceptable, we've had 
a couple phones flat out die, we've had several that had the PoE go 
dead so we were forced to use the AC adapter.  And many of the other 
phones suffer from all kinds of stupid little intermittent issues such 
as dropped calls, reboots and strange ticking and static on the line, 
even on internal calls.  We discovered that quite a few of the network 
cables that came with the phones seemed to be faulty, which when 
replacing them seemed to solve some of our dropped calls and 
spontaneous reboot problems.  Some of the phones had bad handset 
cables.  Replacing some of those seemed to get rid of some of the 
static issues.  We've replaced several of the really troublesome 
phones with Cisco's or Polycom's, and what do you know, no problems 
whatsoever.
 
All in all, we have been extremely disappointed in the reliability of 
the phones and especially unhappy with the level of service from 
Grandstream.  Grandstream should not be permitted to sell these phones 
to consumers, marketed as a business class phone of all things!  
They're knowingly selling a faulty product.  It's outrageous.  And 
what's even more amazing is they claim that the GXP2000 won some sort 
of award, and it's even a "Best Seller" at Atacomm.com 
.  How it the world can that be.  The phone is a 
piece of crap.  They're either going to fix these problems once and 
for all or give us our money back.
 
Here's a good page regarding many of these issues... including the 
echo.. argh...

http://www.voip-info.org/wiki/view/GXP-2000
 
Erick


 
On 12/27/05, [EMAIL PROTECTED] * 
<[EMAIL PROTECTED] > wrote:


>
> What issues are you having with attended call transfer? In
recent months
> I've gone through a fair number of GXP2000 firmware versions and
I can't say
> any of them have had a problem with attended transfer.

I saw this topic and I myself the recommend the same!! Stay away from
Grandstream... unless you are younger, much hair in the head and no
white hair :-)


Isamar

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Re: [Asterisk-Users] Stay away from Grandstream!

[Erick Baum]

We have 50 Grandstream's that we purchased about 3 months ago.  They're all at one site and we've had nothing but trouble with all the phones.  The echo is the worst problem of all.  We had to upgrade from the default 
1.0.1.9 firmware to the 1.0.1.12 "beta" to get the speakerphone to work properly.  Once we did that, there is now a bad echo if one of the GXP users turns their volume up too high, the other party can hear an echo.  If the GXP user turns their volume down a bit, the echo starts to go away.  This happens internally from GXP to GXP as well as with outside callers.  At first Grandstream didn't want to admit that they knew about this or that they'd even ever heard of this problem before.  But they finally at least said they would have an engineer check into it.  Then after some research online I come to find that lots of people are having the echo issue with the newer firmware and they had also notified Grandstream months ago.  So they know about the problem.

 
On top of the echo issue, which is completely unacceptable, we've had a couple phones flat out die, we've had several that had the PoE go dead so we were forced to use the AC adapter.  And many of the other phones suffer from all kinds of stupid little intermittent issues such as dropped calls, reboots and strange ticking and static on the line, even on internal calls.  We discovered that quite a few of the network cables that came with the phones seemed to be faulty, which when replacing them seemed to solve some of our dropped calls and spontaneous reboot problems.  Some of the phones had bad handset cables.  Replacing some of those seemed to get rid of some of the static issues.  We've replaced several of the really troublesome phones with Cisco's or Polycom's, and what do you know, no problems whatsoever.

 
All in all, we have been extremely disappointed in the reliability of the phones and especially unhappy with the level of service from Grandstream.  Grandstream should not be permitted to sell these phones to consumers, marketed as a business class phone of all things!  They're knowingly selling a faulty product.  It's outrageous.  And what's even more amazing is they claim that the GXP2000 won some sort of award, and it's even a "Best Seller" at 
Atacomm.com.  How it the world can that be.  The phone is a piece of crap.  They're either going to fix these problems once and for all or give us our money back.
 
Here's a good page regarding many of these issues... including the echo.. argh... 
http://www.voip-info.org/wiki/view/GXP-2000
 
Erick 
On 12/27/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:

>> What issues are you having with attended call transfer? In recent months> I've gone through a fair number of GXP2000 firmware versions and I can't say
> any of them have had a problem with attended transfer.I saw this topic and I myself the recommend the same!! Stay away fromGrandstream... unless you are younger, much hair in the head and nowhite hair :-)
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Re: [Asterisk-Users] Blackberry SIM card

[Jason p]

BTW chan_bluetooth does not work on the blackberry running 4.0 and
lower , blackberry is locked down so that no calls can be made via
bluetooth (no including the headset) I have read that this is to be
opened up in the new 4.1 code. 
On 12/27/05, Alexander Lopez <[EMAIL PROTECTED]> wrote:
I stand corrected, the 7170 is a BlackBerry that has Wifi Support. TheBES has to support all the BB even if you do not have the option on yourphone chances are that you will have it on the BES.I wanted to get me one a while back but was told that it became
vaporware, I do not think that BB continued development on that product,and lately there attention has been elsewhereThis article is more than a year old and still no BB. I wonder if theCell Carriers got wind of it and shut it down??
http://www.mobiletracker.net/archives/2004/10/18/blackberry_7270.phpAlex> -Original Message-
> From: [EMAIL PROTECTED]> [mailto:[EMAIL PROTECTED]
] On Behalf Of> Kerry Garrison> Sent: Tuesday, December 27, 2005 3:25 PM> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'> Subject: RE: [Asterisk-Users] Blackberry SIM card
>> If you do have the Blackberry Enterprise Server, there are> options available to send out a policy to the devices that> contains SIP server and account information. I have seen no> other way to access those settings nor do I have any clue how
> they would function if I tried to set it to use my Asterisk> server. If I can borrow a Blackberry from my client for a few> days, I will try it.>> Kerry Garrison> Publisher - 
http://GeekGazette.com - http://VOIPSpeak.net> (949) 502-7819 x200 - [EMAIL PROTECTED]> 
http://www.techdatapros.com>>> -Original Message-> From: [EMAIL PROTECTED]> [mailto:
[EMAIL PROTECTED]] On Behalf Of> Alexander Lopez> Sent: Tuesday, December 27, 2005 12:03 PM> To: Asterisk Users Mailing List - Non-Commercial Discussion> Subject: RE: [Asterisk-Users] Blackberry SIM card
>> You do not need the BES server.. It is nice for total> wireless syncronization but not need for it to work.>> The BB will work in three ways:>> BES server, Married with Exchange server or Lotus notes.
>> Internet only, you are given an address like> [EMAIL PROTECTED], you them forward your emails> to it, you can change the way email get sent so it looks like
> your address in stead of [EMAIL PROTECTED] (does not> work to well with Domain Keys or> SPF)>> Redirector, SW is loaded on your machine (PC) and interacts> with Outlook, all mail is sent to your mailbox in an
> encripted file and then the redirector 'redirects it' out to> the user via your PC.>>> As far as Asterisk support goes, the only support that the> BlackBerry has for Asterisk would be the Bluetooth interface.
>> Other nice features that are on but not limited to the BB are:>> Fax to your email as a PDF. (to read you either need a space> lens from the Hubbel Telescope, or a space Electron Microscope!)
>> Your VM left on asterisk include caller information so that> returning calls is as easy as select and click.>>> As far as a SIM card, if your unit is 'unlocked' you can put
> in any GSM Sim card and activate service with whiomever you> fancy, provided they support BlackBerry.> .>>> > -Original Message-> > From: 
[EMAIL PROTECTED]> > [mailto:[EMAIL PROTECTED]] On Behalf Of Kerry> > Garrison> > Sent: Tuesday, December 27, 2005 1:06 PM
> > To: 'Asterisk Users Mailing List - Non-Commercial Discussion'> > Subject: RE: [Asterisk-Users] Blackberry SIM card> >> > To get a sim card you need service from T-Mobile. Any cell
> phone, land> > line, sat phone, etc will work with Asterisk depending on what you> > mean "work with *". You can set up an phone number as a custom> > extension. If you mean you want to setup the blackberry as
> a sip phone> > that communicates with your Asterisk server, I don't really> know how> > you use it within the Blackberry but if you have the Blackberry> > Enterprise Service you can send out a setting to the Blackberry to
> > configure a SIP server. To make this all happen you will need:> >> > Blackberry (you already have that)> > SIM Card (requires T-Mobile service account) Windows> > 2000/2003 Server (maybe you have one, maybe not) Blackberry
> Enterprise> > Server (about $1,500 for a 5 user license)> >> > This is based on my LIMITED experience servicing a client> that has a> > few Blackberrys, I have never touched the SIP settings on theirs, I
> > have only seen them in the BES setup.> > If there is something different that I don't know about,> please let me> > know.> >> > Kerry Garrison> > Publisher - 
http://GeekGazette.com - http://VOIPSpeak.net> > (949) 502-7819 x200 - [EMAIL PROTECTED]
> > http://www.techdatapros.com> >> >> > -Original Message-> > From: 
[EMAIL PROTECTED]> > [mailto:[EMAIL PROTECTED]] On Behalf> Of Robert> > Rawlinson
> > Sent: Tuesday, December 27, 2005 9:50 AM> > To: Asterisk> > Subject: [Asterisk-Users] Blackberry SIM card> >> > I acquired a Blackberry 7100T over Christmas. I had h

[Asterisk-Users] Voicemail as other format?

[Kerry Garrison]

Some users with Blackberry's cant play .wav files, is there a way to save
the voicemail to save as another format like mp3?
-Kerry


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[Asterisk-Users] [Announce] Pending Web-MeetMe update

[Dan Austin]

[New Features]
1.  Added focus to the input textboxes on all pages, so there's one
less mouse click on each page.  Trivial, but something that would
likely trip some users up.

2.  Dynamic generation of year/month/day listboxes to prevent invalid
date selection.  The system still defaults to the current date, but
changing
the month or year will update the day listbox with the correct number of
days.

3.  Added 'Extend' and 'End Now' buttons to the monitor page.  The
'Extend' button adds 10 minutes to the conference.  I have not added
an 'Add Seats' button, and am not sure how critical it is.  I'd like to 
avoid interface clutter if possible.


*
4.  Call history report.  It is now possible to see who was in a
conference
and for how long.  There might be a small issue with this feature, as I
did
not see a clean way to add a 'Back' button.  At the moment the only way
out of this view is to select a menu item from the left side selections.

This functionallity requires a patch to app_meetme to add duration
statistics to the meetme_leave event.  There are a number of options
to accomplish this, but for my environment, this is the best.

It also uses a small PHP script that needs to be running to catch
the manager events.  The script is a little crude at the moment, and
does not deal well if Asterisk not available (crashed/stopped/etc),
but works fine under normal circumstances.

I can either make this feature optional and disabled by default, or
provide my app_meetme patch (all ready in Mantis).  I'd appreciate
comments on which people would prefer.

*

While I am at it, if there happens to be a PHP guru lurking, I'd
appreciate
any comments on cleaning up the code.

I should have the new package ready by the end of the week.

Dan
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[Asterisk-Users] Difference between CDR dispositions..

[El Flynn]

Hi there,

I've got a client complaining about the dispositions in the CDR report we built 
for them:


1. User calls an extension, which rings three SIP phones in the group. Entry in 
extensions.conf:


exten => 100,1,Dial(SIP/200&SIP/201&SIP/202)

2. On three test calls, she dials extension 100 and makes sure no-one picks up 
any of the three phones.


3. In the CDR, two of the calls' disposition are listed as "No Answer", while on 
another CDR entry disposition is listed as "Busy".


Does anyone know what might cause this?

TIA,
Flynn

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RE: [Asterisk-Users] Automatic logoff of all agents at set time

[Bud Bach]

But, if the agents don't log out for some reason, they will still be logged
in the next time the queue opens even if they aren't there right?

-- Bud

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Michiel van Baak
> Sent: Tuesday, December 27, 2005 5:27 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Automatic logoff of all agents at set time
> 
> On 16:22, Tue 27 Dec 05, Chuck Bunn wrote:
> > Hi,
> >
> > Is there a way to force the logoff of all agents at a set time say
> > 8:00PM or do I have to do an agentcallbacklogin with each agents
> > credentials? I am using Asterisk 1.2 The wiki shows an extension that
> > the agent calls to preform the logoff - I need something that is
> > completely automated as we need calls to stop going to a queue and to go
> > to voice mail after hours.
> >
> 
> Hi,
> 
> You dont have to logoff your agents to do this.
> Have a look at the extensions.conf cmd GotoIfTime:
> http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime
> 
> Good luck
> 
> --
> Michiel van Baak
> http://michiel.vanbaak.info
> [EMAIL PROTECTED]
> GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D
> 
> "Why is it drug addicts and computer afficionados are both called users?"
> 
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Re: [Asterisk-Users] Automatic logoff of all agents at set time

[Michiel van Baak]

On 16:54, Tue 27 Dec 05, Chuck Bunn wrote:
> Hi,
> 
> I understand how GototIfTime works but that still leaves agents logged 
> in and if an agent is absent the next day calls will go to an agent that 
> is not there.

Hi,

That is just how you educate your agents.
We found out that a penalty works great.
Simply tell them it's part of their job to logoff when they
leave. If not, keep 100 us $ from their paycheck.
Works like a charm.

-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

"Why is it drug addicts and computer afficionados are both called users?"

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Re: [Asterisk-Users] Automatic logoff of all agents at set time

[Chuck Bunn]

Hi,

I understand how GototIfTime works but that still leaves agents logged 
in and if an agent is absent the next day calls will go to an agent that 
is not there.


Thanks

Michiel van Baak wrote:


On 16:22, Tue 27 Dec 05, Chuck Bunn wrote:
 


Hi,

Is there a way to force the logoff of all agents at a set time say 
8:00PM or do I have to do an agentcallbacklogin with each agents 
credentials? I am using Asterisk 1.2 The wiki shows an extension that 
the agent calls to preform the logoff - I need something that is 
completely automated as we need calls to stop going to a queue and to go 
to voice mail after hours.


   



Hi,

You dont have to logoff your agents to do this.
Have a look at the extensions.conf cmd GotoIfTime:
http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime

Good luck

 



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Re: [Asterisk-Users] Automatic logoff of all agents at set time

[Michiel van Baak]

On 16:22, Tue 27 Dec 05, Chuck Bunn wrote:
> Hi,
> 
> Is there a way to force the logoff of all agents at a set time say 
> 8:00PM or do I have to do an agentcallbacklogin with each agents 
> credentials? I am using Asterisk 1.2 The wiki shows an extension that 
> the agent calls to preform the logoff - I need something that is 
> completely automated as we need calls to stop going to a queue and to go 
> to voice mail after hours.
> 

Hi,

You dont have to logoff your agents to do this.
Have a look at the extensions.conf cmd GotoIfTime:
http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime

Good luck

-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

"Why is it drug addicts and computer afficionados are both called users?"

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[Asterisk-Users] Automatic logoff of all agents at set time

[Chuck Bunn]

Hi,

Is there a way to force the logoff of all agents at a set time say 
8:00PM or do I have to do an agentcallbacklogin with each agents 
credentials? I am using Asterisk 1.2 The wiki shows an extension that 
the agent calls to preform the logoff - I need something that is 
completely automated as we need calls to stop going to a queue and to go 
to voice mail after hours.


Thanks
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Re: [Asterisk-Users] PRI: This number has been disconnected

[Francesco Peeters (Asterisk)]

On Tue, December 27, 2005 23:37, Javier Ergas said:
> Hi,
>
>
>
> I'm running [EMAIL PROTECTED] 1.5 with TE110P E1 PRI in Chile.
>
> When calling an invalid number using, I expect to hear:
>
> "We're sorry you have reached a number which has been disconnected ..."
>
> And that is indeed what I hear when I dial out from [*] using analog FXO,
> or
> VoicePulse or NuPhone.  When I dial that same number trough the T1 / PRI
> interface however, I only hear the allison7/all-circuits-busy-now message.
>
>
>
> There was another issue like this in an old post
> (http://lists.digium.com/pipermail/asterisk-users/2004-April/043597.html)
> but I think it isn't the same.
>
>


I believe this has to do with the AMP macro's being used in [EMAIL PROTECTED] I 
am
seeing similar things.

For instance: One issue I have is that when a route has multiple trunks,
and the first trunk after a while returns with 'NOANSWER', it merrily
continues to the next trunk, which is not quite the behavior I'd expect.
Especially as the primary trunk (IAX/VoipBuster) is *much* cheaper (ie
free) as compared to the second trunk (Zap/g1), but the switch is made
without any message. This could mean that you might be talking to someone
on a different trunk, and instead of a free call, be paying normal fees.

This could become expensive if you're calling the USA from Europe!...

I am currently looking in to ways to enhance those macro's to respond more
reliably, as well as return more useful information (busy tone on busy and
no-answer, number disconnected info, etc.) when needed.

If I do get to a satifactory set of macro's, I will put them up on the
Wiki and let the list know... (I'm just starting on doing manual
configuring, so it will be a tough job to crack, but also a learning
experience...)

-- 
F Peeters
  PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch
  2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0
Cologne HFC-S pins #52, #54, #55 connected in parallel for synching.
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RE: [Asterisk-Users] Blackberry SIM card

[Alexander Lopez]

I stand corrected, the 7170 is a BlackBerry that has Wifi Support. The
BES has to support all the BB even if you do not have the option on your
phone chances are that you will have it on the BES.

I wanted to get me one a while back but was told that it became
vaporware, I do not think that BB continued development on that product,
and lately there attention has been elsewhere

This article is more than a year old and still no BB. I wonder if the
Cell Carriers got wind of it and shut it down??
 
http://www.mobiletracker.net/archives/2004/10/18/blackberry_7270.php



Alex

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Kerry Garrison
> Sent: Tuesday, December 27, 2005 3:25 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Blackberry SIM card
> 
> If you do have the Blackberry Enterprise Server, there are 
> options available to send out a policy to the devices that 
> contains SIP server and account information. I have seen no 
> other way to access those settings nor do I have any clue how 
> they would function if I tried to set it to use my Asterisk 
> server. If I can borrow a Blackberry from my client for a few 
> days, I will try it.
> 
> Kerry Garrison
> Publisher - http://GeekGazette.com - http://VOIPSpeak.net
> (949) 502-7819 x200 - [EMAIL PROTECTED] 
> http://www.techdatapros.com 
>  
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Alexander Lopez
> Sent: Tuesday, December 27, 2005 12:03 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Blackberry SIM card
> 
> You do not need the BES server.. It is nice for total 
> wireless syncronization but not need for it to work. 
> 
> The BB will work in three ways:
> 
> BES server, Married with Exchange server or Lotus notes. 
> 
> Internet only, you are given an address like 
> [EMAIL PROTECTED], you them forward your emails 
> to it, you can change the way email get sent so it looks like 
> your address in stead of [EMAIL PROTECTED] (does not 
> work to well with Domain Keys or
> SPF)
> 
> Redirector, SW is loaded on your machine (PC) and interacts 
> with Outlook, all mail is sent to your mailbox in an 
> encripted file and then the redirector 'redirects it' out to 
> the user via your PC. 
> 
> 
> As far as Asterisk support goes, the only support that the 
> BlackBerry has for Asterisk would be the Bluetooth interface.
> 
> Other nice features that are on but not limited to the BB are:
> 
> Fax to your email as a PDF. (to read you either need a space 
> lens from the Hubbel Telescope, or a space Electron Microscope!)
> 
> Your VM left on asterisk include caller information so that 
> returning calls is as easy as select and click.
> 
> 
> As far as a SIM card, if your unit is 'unlocked' you can put 
> in any GSM Sim card and activate service with whiomever you 
> fancy, provided they support BlackBerry.
> .
>  
> 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Kerry 
> > Garrison
> > Sent: Tuesday, December 27, 2005 1:06 PM
> > To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> > Subject: RE: [Asterisk-Users] Blackberry SIM card
> > 
> > To get a sim card you need service from T-Mobile. Any cell 
> phone, land 
> > line, sat phone, etc will work with Asterisk depending on what you 
> > mean "work with *". You can set up an phone number as a custom 
> > extension. If you mean you want to setup the blackberry as 
> a sip phone 
> > that communicates with your Asterisk server, I don't really 
> know how 
> > you use it within the Blackberry but if you have the Blackberry 
> > Enterprise Service you can send out a setting to the Blackberry to 
> > configure a SIP server. To make this all happen you will need:
> > 
> > Blackberry (you already have that)
> > SIM Card (requires T-Mobile service account) Windows
> > 2000/2003 Server (maybe you have one, maybe not) Blackberry 
> Enterprise 
> > Server (about $1,500 for a 5 user license)
> > 
> > This is based on my LIMITED experience servicing a client 
> that has a 
> > few Blackberrys, I have never touched the SIP settings on theirs, I 
> > have only seen them in the BES setup.
> > If there is something different that I don't know about, 
> please let me 
> > know.
> > 
> > Kerry Garrison
> > Publisher - http://GeekGazette.com - http://VOIPSpeak.net
> > (949) 502-7819 x200 - [EMAIL PROTECTED] 
> > http://www.techdatapros.com
> >  
> > 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf 
> Of Robert 
> > Rawlinson
> > Sent: Tuesday, December 27, 2005 9:50 AM
> > To: Asterisk
> > Subject: [Asterisk-Users] Blackberry SIM card
> > 
> > I acquired a Blackberry 7100T over Christmas. I had heard 
> it will work 
> > with
> > * and that is what I want to do with it. But I think it needs a SIM 
> > ca

RE: [Asterisk-Users] Blackberry SIM card

[Chris Bagnall]

(Bear in mind my knowledge of Blackberry stuff is UK-based, not US-based.
How different it is over there I don't know)

> To get a sim card you need service from T-Mobile.

The Blackberry devices are no different from any other GSM cellphone, so you
can get a SIM card from anyone, not specifically T-Mobile (provided your
device hasn't been locked to a network in the first place, and if it has,
there are plenty of market traders who'll unlock the thing for a few
pounds).

> BES server, Married with Exchange server or Lotus notes. 
> Internet only, you are given an address like 
> [EMAIL PROTECTED], you them forward your emails 
> to it, you can change the way email get sent so it looks like 
> your address in stead of [EMAIL PROTECTED] (does not 
> work to well with Domain Keys or
> SPF)

The Blackberry will also work fine with standard IMAP/POP3/SMTP servers, so
you aren't tied to an email address they provide in any way/shape/form.

I must point out I've no experience using the SIP side of the device at all.
I might try and steal one off a client in the new year and have a play with
it - I'd no idea it supported SIP at all.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


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Re: [Asterisk-Users] Asterisk Hosting

[Michiel van Baak]

On 14:50, Tue 27 Dec 05, BILL GITONGA wrote:
> What is the best method of storing voice main messages
> so that they are accessible to different asterisk
> servers in a hosted environment? I have considered
> Asterisk real time but I don’t think it stores the
> actual voice mail folder in the database. I’m thinking
> of using NFS for this and put my voice mail folders on
> the NFS so that it is accessible by the different
> servers. Is this a good way to do it or is there a
> better way of doing this?
> 

If you want it all in a database you can try odbc.
My opinion is a database is not for binary files, so what I
do is use NFS. Works great.
-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

"Why is it drug addicts and computer afficionados are both called users?"

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[Asterisk-Users] Asterisk Hosting

[BILL GITONGA]

What is the best method of storing voice main messages
so that they are accessible to different asterisk
servers in a hosted environment? I have considered
Asterisk real time but I don’t think it stores the
actual voice mail folder in the database. I’m thinking
of using NFS for this and put my voice mail folders on
the NFS so that it is accessible by the different
servers. Is this a good way to do it or is there a
better way of doing this?




__ 
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http://brand.yahoo.com/cybergivingweek2005/
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[Asterisk-Users] PRI: This number has been disconnected

[Javier Ergas]

Hi,

 

I’m running [EMAIL PROTECTED] 1.5 with TE110P E1
PRI in Chile.

When calling an invalid number using, I expect to
hear:

"We're sorry you have reached a number which has
been disconnected ..."

And that is indeed what I hear when I dial out from
[*] using analog FXO, or VoicePulse or NuPhone.  When I dial that same
number trough the T1 / PRI interface however, I only hear the allison7/all-circuits-busy-now
message.

 

There was another issue like this in an old post (http://lists.digium.com/pipermail/asterisk-users/2004-April/043597.html)
but I think it isn’t the same.

 

Any ideas??

 

Thanks,


Javier Ergas

CEO

Pibix.cl

 

Here is the trace from the Telcom:

 

PROTOCOL DISCRIMINATOR: 8 Q.931 USER-NETWORK CALL
CONTROL MESSAGES

CALL REFERENCE LENGTH:2 FLAG:0(Orig) VALUE:418

M   0x05 SETUP  Length: 35

I   0x04 BEARER CAPABILITY Length: 3

3   0x80 Coding
standard:  -00- CCITT standard

 Inform.transf.cap.:  
---0 speech

4   0x90 Transfer
mode:    -00- circuit mode


Inform.transf.rate:   ---1 64 kbits/s

5   0xA3 Layer 1
ident.:   -01- 

 User
info layer 1:    ---00011 recomm. G.711 A-law

I   0x18 CHANNEL IDENTIFICATION Length: 3

3   0xA9 Interface id. pres.:  -0--
interface implicitly identifier

 Interface
type:   --1- other interface (primary)

 Spare
---0 


Preferred/exclusive:  1--- exclusive: only the indicated channel is
acceptable

 D-channel
indicator:  -0-- the channel identified is not the D-channel

 Channel
selection:    --01 as indicated in following octets

3.2 0x83 Coding
standard:  -00- CCITT standard


Number/Map:  
---0 channel is indicated by number in the following octet

 Channel
type: 0011 B-channel units


Channel number: Length: 1

  
VALUE: 7

I   0x28 DISPLAY Length: 4

 Display
information: Length: 4

  
CONTENT: test

I   0x6C CALLING PARTY NUMBER Length: 6

3   0x21 Type of
number:   -010 national number

 Numbering
plan id.:   0001 ISDN/Telephony numbering plan

3a  0x80 Presentation ind.:   
-00- allowed


Spare
---000-- 

 Screening
ind.:   --00 user-provided, not screened

 CLI:
Length: 4

  
CONTENT: 9349

I   0x70 CALLED PARTY NUMBER Length: 8

3   0x80 Type of
number:   -000 unknown

     Numbering
plan id.:    unknown

 Called:
Length: 7

  
CONTENT: 2514990

I   0xA1 SENDING COMPLETE

1   0xA1 Info. el.
id.:    1011 

 

 

PROTOCOL DISCRIMINATOR: 8 Q.931 USER-NETWORK CALL
CONTROL MESSAGES

CALL REFERENCE LENGTH:2 FLAG:1(Dest) VALUE:418

M   0x7D STATUS  Length: 9

I   0x08 CAUSE Length: 4

3   0x82 Coding
standard:  -00- CCITT standard


Spare
---0 

 Location:
0010 public network serving the local user

4   0xE3 Cause
value:  -1100011 D 99:
Information element non-existent or not implemented

 Diagnostics:
Length: 2

    98h 28h 

I   0x14 CALL STATE Length: 1

3   0x01 Coding
standard:  00-- CCITT standard

 Call
state/glob int:  --01 call Initiated

 

 

PROTOCOL DISCRIMINATOR: 8 Q.931 USER-NETWORK CALL
CONTROL MESSAGES

CALL REFERENCE LENGTH:2 FLAG:1(Dest) VALUE:418

M   0x02 CALL PROCEED  Length: 5

I   0x18 CHANNEL IDENTIFICATION Length: 3

3   0xA9 Interface id. pres.:  -0--
interface implicitly identifier

 Interface
type:   --1- other interface (primary)

 Spare
---0 


Preferred/exclusive:  1--- exclusive: only the indicated channel is
acceptable

 D-channel
indicator:  -0-- the channel identified is not the D-channel

 Channel
selection:    --01 as indicated in following octets

3.2 0x83 Coding
standard:  -00- CCITT standard


Number/Map:  
---0 channel is indicated by number in the following octet

 Channel
type: 0011 B-channel units


Channel number: Length: 1

  
VALUE: 7

 

 

PROTOCOL DISCRIMINATOR: 8 Q.931 USER-NETWORK CALL
CONTROL MESSAGES

CALL REFERENCE LENGTH:2 FLAG:1(Dest) VALUE:418

M   0x45 DISCONNECT  Length: 12

I   0x08 CAUSE Length: 2

3   0x80 Coding
standard:  -00- CCITT standard


Spare
---0 


Location:
 user

4   0x81 Cause
value:  -001 D
1:Unallocated number

I   0x1E PROGRESS INDICATOR Length: 2

3   0x82 Coding
standard:  -00- CCITT standard


Spare
---0 

 Location:
0010 public network serving the local user

4   0x88 Progress
desc.:   -0001000 in-band information or
appropriate pattern now available

I   0x1E PROGRESS INDICATOR Length: 2

3   0x82 Coding
standard:  -00- CCITT s

Re: [Asterisk-Users] polycom sip slower than grandstream

[Jerry Jones]

Its all in the priority the phones assign responding to messages.  
Sipura are also fast compared to Polycom.


this is not an accurate measure of network latency.


On Dec 27, 2005, at 2:11 PM, Dean Collins wrote:

I have a polycom 501, for some reason asterisk always shows the  
round trip time to it as being significantly higher than the 2  
grandstreams, all 3 are on the same lan.








Grandstream 40/40192.168.16.40D   
255.255.255.255  5060 OK (4 ms)


Grandstream 31/31192.168.16.31D   
255.255.255.255  5060 OK (4 ms)


Polycom   30/30192.168.16.30D   
255.255.255.255  5060 OK (78 ms)








Any thoughts?





Dean

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Re: [Asterisk-Users] UK, Disconnect supervision

[Jonathan Attwood]

Which firmware version are you using on your spa3000?

Peter Hoppe wrote:
|| Hello!
|| 
|| This is actually less a question than some information, if anyone else

|| struggles with the same issue.
|| 
|| I am located in the UK and use a Sipura-3000 adapter to connect to a BT

|| line (via fxo port). One problem I had was that disconnect supervision
|| didn't work:
|| 
|| Some caller phones me (my adapter)

|| adapter goes off-hook (answers call)
|| caller hangs up
|| adapter doesn't realize and stays off hook.
|| 
|| I researched into it and found that the BT exchange delivers a CPC (0.1

|| sec) upon the caller's hangup and a disconnect tone about 3 seconds
|| later. The tone lasts 6 seconds and remains constant during that time.
|| I recorded the disconnect tone with Cool Edit and did a frequency
|| analysis on it and got the following components:
|| 
|| 400Hz/-56dB +

|| 1200Hz/-69dB +
|| 2000Hz/-65dB +
|| 2800Hz/-59dB +
|| 3600Hz/-59dB
|| 
|| If normalized with -56 dB reference level, I get
|| 
|| 400Hz/0dB +

|| 1200Hz/-13dB +
|| 2000Hz/-9dB +
|| 2800Hz/-3dB +
|| 3600Hz/-3dB
|| 
|| I suspect that the harmonics (1200, 2000, 2800, 3600)Hz may not have

|| come from the Exchange, but were distortions.
|| 
|| After the 6 seconds the tone stopped and there was succession of two

|| 'clicks' on the line (another CPC? haven't looked into that); both were
|| 0.3 secs apart.
|| 
|| I tried to use both, the CPC and the disconnect tone in the Sipura-3000

|| settings. Unfortunately I couldn't get disconnect supervision via the
|| disconnect tone to work. In a post on the voxilla forum
|| 
|| http://voxilla.com/PNphpBB2-viewtopic-t-2904.html
|| 
|| I found some further info on disconnect tone in the UK - unfortunately

|| I couldn't get those settings to work. I also looked into BT's SIN
|| notes (on http://www.sinet.bt.com , notes 350, 351) but failed to see
|| further information on what tone they exactly deliver for a disconnect
|| event. 
|| 
|| However, the CPC did work. I set the adapter to a CPC minimum time of

|| 0.05 seconds, and from then on it recognized remote disconnection in
|| every test phone call. If I set the time too short I got problems - for
|| incoming calls the adapter started to mistake the incoming ringing for
|| disconnect events and simply wouldn't go off hook anymore.
|| 
|| Now, these are the settings I configured my adapter with:
|| 
|| Settings:

|| spa-3000-setup-web-page/PSTN-line-tab/PSTN-disconnect-detection-section:
|| 
|| 'Detect CPC' => 'yes'

|| 'Min CPC Duration' => '0.05'
|| 
|| I also have

|| 'Detect Disconnect Tone' => 'no' [as it didn't work]
|| 'Disconnect Tone' => '[EMAIL PROTECTED],[EMAIL PROTECTED];5(5/5/1+2)'
|| 'Detect Polarity Reversal' => 'Yes'
|| 'Detect PSTN Long Silence' => 'yes'
|| 'PSTN Long Silence Duration' => '150'
|| 'Detect VoIP Long Silence' => 'no'
|| 'VoIP Long Silence Duration' => '30'
|| 
|| but the relevant values are 'Detect CPC' => 'yes' and 'Min CPC

|| Duration' => '0.05'
|| 
|| 
|| I hope this helps, if anyone struggles with unrecognized disconnects.
|| 
|| God bless, Peter
|| 
|| 
|| CPC: Calling Party Control: A short break in the line current in the

|| called party's phone line when the calling party hangs up.
|| (def. from http://www.vikingelectronics.com/glossary/telecom-term.php)
|| --
|| dyslexics of the world - untie !
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RE: [Asterisk-Users] spandsp & fax

[Carlos Alperin]

Don,

The previous question I believe was what linux are you using?

By the way, I would like to know that too, just I was trying to make this
work for weeks with no success.

Thanks,

Carlos Alperin


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dov Bigio
Sent: Tuesday, December 27, 2005 10:54 AM
To: Kristof Hardy; Asterisk Users Mailing List - Non-CommercialDiscussion
Subject: Re: [Asterisk-Users] spandsp & fax

Hi BJ, Kristof,

It worked!

I am using the version at
http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.2pre21c/asterisk-1.
2.x/.

I think I had bad symlinks on /usr/local/lib and by reading the tutorial on
AsteriskGuru I found that... (The previously installed version of spandsp
has been 0.0.3, but now you have installed version 0.0.2. The problem is
that the installation of version 0.0.3 creates a symlink, which is not
replaced by installation of version 0.0.2. So the symlink points to the
library of version 0.0.3, which actually does not exist.). I simply deleted
all files related to spandsp from this directory and installed it again!

Thank you
Dov


- Original Message - 
From: "Kristof Hardy" <[EMAIL PROTECTED]>
To: "Dov Bigio" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List -
Non-CommercialDiscussion" 
Sent: Tuesday, December 27, 2005 12:59 PM
Subject: Re: [Asterisk-Users] spandsp & fax


> Dov Bigio wrote:
> > I am using Asterisk 1.2.1 and followed instructions on
> > http://www.asteriskguru.com/tutorials/spandsp.html to install faxing
> > capability on my server.
>
> what platform are you running on? (wich distro?)
> Does the make of the app_txfax and app_rxfax work out well?
>
>
>
>


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Re: [Asterisk-Users] dtmf problem

[Janina Sajka]

[EMAIL PROTECTED] writes:
> Bart,
> 
> We have has similar issues with BroadVoice in the past. From what I
> understand they had problems with DTMF depending on which proxy you register
> to. This is a bug that related to their session border controllers which
> should have been resolved.
> 
... snip snip ...
> 
> 4. Change your DTMF mode from inband to rfc2833. BroadVoice does support
> out-of-band DTMF signalling, though their website is out of date.
> 
> i.e. dtmfmode=rfc2833

I can get dtmf on outbound calls with rfc2833 using either the nyc or
dca Broadvoice servers, but I can only get dtmf working for incoming
calls if I leave it as inband.

Janina

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Re: [Asterisk-Users] How to check Digium TE410P firmware version?

[C F]

Use dmesg

On 12/27/05, Franz Wu <[EMAIL PROTECTED]> wrote:
> I connect to Asterisk via SSH all the times. Did not notice about console
> messages about module loading.
>
> Thanks
>
> - Original Message -
> From: "BJ Weschke" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Tuesday, December 27, 2005 10:24
> Subject: Re: [Asterisk-Users] How to check Digium TE410P firmware version?
>
>
> On 12/26/05, Franz Wu <[EMAIL PROTECTED]> wrote:
> > Hi list
> > I have one TE410P and want to know how to. Sending back to Digium should
> > be
> > a good idea.
> >
>
>  When you load the wct4xxp module you'll see (1st Gen) or (2nd Gen)
> pop up which will indicate which version of the firmware the board is.
>
> --
> Bird's The Word Technologies, Inc.
> http://www.btwtech.com/
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Re: [Asterisk-Users] Play soundfile before snswer

[C F]

Kerry, the OP wanted to know if it's possible to do so so that billing
doesn't start. If you call a toll free number from overseas then you
will hear a recording telling you something like this: you will be
charged long distance charges if you continue this call.
You shouldn't be charged if you hangup while it's playing, and that is
the same as using noanswer option in playback.

On 12/27/05, Kerry Garrison <[EMAIL PROTECTED]> wrote:
> How would you play a file to a line that hasn't been picked up? You have to
> pick up the line in order to do anything with it.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Arik Funke
> Sent: Tuesday, December 27, 2005 1:33 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Play soundfile before snswer
>
> Hello,
>
> can anybody tell me, if it is possible to play a soundfile to a caller
> BEFORE having picked up? Will the call be billed for the caller on PSTN?
>
> Best regards,
> Arik
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Re: [Asterisk-Users] Play soundfile before snswer

[C F]

If you have a PRI you can use app_playback with the NOANSWER option,
check show application playback in the CLI

On 12/27/05, Arik Funke <[EMAIL PROTECTED]> wrote:
> Hello,
>
> can anybody tell me, if it is possible to play a soundfile to a caller
> BEFORE having picked up? Will the call be billed for the caller on PSTN?
>
> Best regards,
> Arik
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Re: [Asterisk-Users] Asterisk on VPS

[Blake OPS]

On 12/27/05, Ross C <[EMAIL PROTECTED]> wrote:
> I'm curious if anyone has tried installing Asterisk on a Virtual Private
> Server from a web hosting company?  I am a web hosting reseller with
> Jodohost.com, so I can have as many Linux VPS's as I want, and I thought I
> might try it.  I'm just curious if anyone else has tried this before?

Da Beave from Telephreak wrote a paper on this. Check it out at
http://www.telephreak.org/papers/vpa/

-BlakeOPS
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Re: [Asterisk-Users] asterisk AVM C2 again

[Armin Schindler]

On Tue, 27 Dec 2005, Dave Cotton wrote:
> On Tue, 2005-12-27 at 19:27 +0100, Armin Schindler wrote:
> 
> > It looks like the call is signaled on both ports !?
> 
> On another installation in France I'm also getting this, but with 2
> Fritz! cards, the call is signalled on both cards.

Is this some feature of the line configuration/protocol?
I never heard of this before.

Armin

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RE: [Asterisk-Users] Play soundfile before snswer

[Kerry Garrison]

How would you play a file to a line that hasn't been picked up? You have to
pick up the line in order to do anything with it. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Arik Funke
Sent: Tuesday, December 27, 2005 1:33 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Play soundfile before snswer

Hello,

can anybody tell me, if it is possible to play a soundfile to a caller
BEFORE having picked up? Will the call be billed for the caller on PSTN?

Best regards,
Arik
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[Asterisk-Users] Play soundfile before snswer

[Arik Funke]

Hello,

can anybody tell me, if it is possible to play a soundfile to a caller 
BEFORE having picked up? Will the call be billed for the caller on PSTN?


Best regards,
Arik
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Re: [Asterisk-Users] asterisk AVM C2 again

[Dave Cotton]

On Tue, 2005-12-27 at 19:27 +0100, Armin Schindler wrote:

> It looks like the call is signaled on both ports !?

On another installation in France I'm also getting this, but with 2
Fritz! cards, the call is signalled on both cards.

-- 
Dave Cotton <[EMAIL PROTECTED]>

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[Asterisk-Users] Polycom Soundpoint 501 outbound calls always show NO ANSWER

[Reid W. Johnson]

Hi everyone,

I just upgraded my Asterisk box from 1.0 to 1.2, immediately after the
upgrade my Polycom Soundpoint 501 stop working. All outbound calls from
my phone show "NO ANSWER" in the CDR, the call connects but disconnects
after 60 second. Inbound calls to this phone work perfectly and all of
my other phone are working properly both inbound and outbound. Any
ideas?

Thanks in advance.
Reid


 Here are specs for my system:

CentOS 4.2
Asterisk SVN-branch-1.2-r7497
Digium TDM400P 2 FXO, 1FXS
Polycom Soundpoint 501

Here is an entry for a failed outbound call:

Asterisk SVN-branch-1.2-r7497 built by root @ asterisk.corenetwork.ca on
a i686 running Linux on 2005-12-27 20:03:51 UTC
-- Executing Dial("SIP/rjohnsonhs-64ba", "Zap/2/") in new
stack
-- Called 2/
-- Zap/2-1 is ringing
-- Hungup 'Zap/2-1'
  == Spawn extension (internal, , 1) exited non-zero on
'SIP/rjohnsonhs-9994'

Here is an inbound call to the same phone:

  == Spawn extension (internal, xxx, 1) exited non-zero on
'SIP/rjohnsonhs-9994'
-- Starting simple switch on 'Zap/2-1'
-- Executing Wait("Zap/2-1", "2") in new stack
-- Executing Answer("Zap/2-1", "") in new stack
-- Executing DigitTimeout("Zap/2-1", "10") in new stack
-- Set Digit Timeout to 10
-- Executing ResponseTimeout("Zap/2-1", "15") in new stack
-- Set Response Timeout to 15
-- Executing GotoIfTime("Zap/2-1",
"9:00-18:00|mon-fri|*|*?incoming|s|9") in new stack
-- Goto (incoming,s,9)
-- Executing BackGround("Zap/2-1", "cns-announce") in new stack
-- Playing 'cns-announce' (language 'en')
  == CDR updated on Zap/2-1
-- Executing Goto("Zap/2-1", "RJohnson|s|1") in new stack
-- Goto (RJohnson,s,1)
-- Executing Playback("Zap/2-1", "cns-ip-transfer") in new stack
-- Playing 'cns-ip-transfer' (language 'en')
-- Executing DigitTimeout("Zap/2-1", "5") in new stack
-- Set Digit Timeout to 5
-- Executing ResponseTimeout("Zap/2-1", "10") in new stack
-- Set Response Timeout to 10
-- Executing Wait("Zap/2-1", "1") in new stack
-- Executing Dial("Zap/2-1", "SIP/rjohnsonhs|15|m") in new stack
-- Called rjohnsonhs
-- Started music on hold, class 'default', on channel 'Zap/2-1'
-- SIP/rjohnsonhs-6d0a is ringing
-- SIP/rjohnsonhs-6d0a answered Zap/2-1
-- Stopped music on hold on Zap/2-1
  == Spawn extension (RJohnson, s, 5) exited non-zero on 'Zap/2-1'
-- Hungup 'Zap/2-1'
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[Asterisk-Users] How to register a sip user/peer in real time

[Rehan Ahmed]

Hello
 
Can some one point me to more info on how to register a SIP PEER or a user on asterisk, say a FWD account on Real time database.
 
Thank You
 
Rehan-- Rehan Ahmed AllahWalahttp://www.SuperTec.com - Tommrow's Technology, Today.http://www.didx.net - DID Number Exchange and Peering Service.
 
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Re: [Asterisk-Users] How to check Digium TE410P firmware version?

[Franz Wu]

I connect to Asterisk via SSH all the times. Did not notice about console 
messages about module loading.

Thanks

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

Sent: Tuesday, December 27, 2005 10:24
Subject: Re: [Asterisk-Users] How to check Digium TE410P firmware version?


On 12/26/05, Franz Wu <[EMAIL PROTECTED]> wrote:
> Hi list
> I have one TE410P and want to know how to. Sending back to Digium should 
> be
> a good idea.
>

 When you load the wct4xxp module you'll see (1st Gen) or (2nd Gen)
pop up which will indicate which version of the firmware the board is.

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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[Asterisk-Users] Asterisk Realtime Database Redundancy

[Douglas Garstang]

In short, does Asterisk have any database redundancy???

Is there any way to specific more than one db host in res_mysql.conf? If you 
specify dbhost with a hostname, and use round-robin dns, does Asterisk read 
this file only on startup or on every db connect attempt? If it fails to get a 
connection, is it start enough to try the next host?

What about with the MySQL application? Is there any way to incorporate 
redundancy into this, such that it can try multiple db servers when one is 
unavailable? Is there any way around the limitation where Asterisk reads 
extensions.conf on startup, and if that db goes down, Asterisk has already 
loaded the IP address into memory... and your screwed?

And what about Realtime extensions? Is there any database redundancy there at 
all? What happens when this happens:

*CLI> [Dec 27 06:18:15] ERROR[3007]: res_config_mysql.c:615 mysql_reconnect: 
MySQL RealTime: Failed to connect database server vox180 on 192.168.10.15. 
Check debug for more info.
[Dec 27 06:18:15] ERROR[3007]: res_config_mysql.c:615 mysql_reconnect: MySQL 
RealTime: Failed to connect database server vox180 on 192.168.10.15. Check 
debug for more info.
[Dec 27 06:18:15] ERROR[3007]: res_config_mysql.c:615 mysql_reconnect: MySQL 
RealTime: Failed to connect database server vox180 on 192.168.10.15. Check 
debug for more info.
[Dec 27 06:18:15] ERROR[3007]: res_config_mysql.c:615 mysql_reconnect: MySQL 
RealTime: Failed to connect database server vox180 on 192.168.10.15. Check 
debug for more info.

Is Asterisk smart enough to be able to try another database? It looks like even 
when there are following contexts with a number, Asterisk immediately fails the 
call at this point.

Thanks,
Douglas.
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[Asterisk-Users] agent logs

[Hall, Eric M.]

I'm looking for a ay 
to track when an agent logs in and logs out. Best if it could be put in a 
mysql db but a text file will be ok for now..
 
 
Any help would  
be great !
 
 
Thanks
 
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RE: [Asterisk-Users] Blackberry SIM card

[Kerry Garrison]

If you do have the Blackberry Enterprise Server, there are options available
to send out a policy to the devices that contains SIP server and account
information. I have seen no other way to access those settings nor do I have
any clue how they would function if I tried to set it to use my Asterisk
server. If I can borrow a Blackberry from my client for a few days, I will
try it.

Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com 
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander
Lopez
Sent: Tuesday, December 27, 2005 12:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Blackberry SIM card

You do not need the BES server.. It is nice for total wireless
syncronization but not need for it to work. 

The BB will work in three ways:

BES server, Married with Exchange server or Lotus notes. 

Internet only, you are given an address like
[EMAIL PROTECTED], you them forward your emails to it, you can
change the way email get sent so it looks like your address in stead of
[EMAIL PROTECTED] (does not work to well with Domain Keys or
SPF)

Redirector, SW is loaded on your machine (PC) and interacts with Outlook,
all mail is sent to your mailbox in an encripted file and then the
redirector 'redirects it' out to the user via your PC. 


As far as Asterisk support goes, the only support that the BlackBerry has
for Asterisk would be the Bluetooth interface.

Other nice features that are on but not limited to the BB are:

Fax to your email as a PDF. (to read you either need a space lens from the
Hubbel Telescope, or a space Electron Microscope!)

Your VM left on asterisk include caller information so that returning calls
is as easy as select and click.


As far as a SIM card, if your unit is 'unlocked' you can put in any GSM Sim
card and activate service with whiomever you fancy, provided they support
BlackBerry.
.
 

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Kerry 
> Garrison
> Sent: Tuesday, December 27, 2005 1:06 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Blackberry SIM card
> 
> To get a sim card you need service from T-Mobile. Any cell phone, land 
> line, sat phone, etc will work with Asterisk depending on what you 
> mean "work with *". You can set up an phone number as a custom 
> extension. If you mean you want to setup the blackberry as a sip phone 
> that communicates with your Asterisk server, I don't really know how 
> you use it within the Blackberry but if you have the Blackberry 
> Enterprise Service you can send out a setting to the Blackberry to 
> configure a SIP server. To make this all happen you will need:
> 
> Blackberry (you already have that)
> SIM Card (requires T-Mobile service account) Windows
> 2000/2003 Server (maybe you have one, maybe not) Blackberry Enterprise 
> Server (about $1,500 for a 5 user license)
> 
> This is based on my LIMITED experience servicing a client that has a 
> few Blackberrys, I have never touched the SIP settings on theirs, I 
> have only seen them in the BES setup.
> If there is something different that I don't know about, please let me 
> know.
> 
> Kerry Garrison
> Publisher - http://GeekGazette.com - http://VOIPSpeak.net
> (949) 502-7819 x200 - [EMAIL PROTECTED] 
> http://www.techdatapros.com
>  
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Robert 
> Rawlinson
> Sent: Tuesday, December 27, 2005 9:50 AM
> To: Asterisk
> Subject: [Asterisk-Users] Blackberry SIM card
> 
> I acquired a Blackberry 7100T over Christmas. I had heard it will work 
> with
> * and that is what I want to do with it. But I think it needs a SIM 
> card to make it work. If this is true how do I go about getting a SIM 
> card for it and how to set it up? Thanks for any help you can offer.
> Bob Rawlinson
> 
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RE: [Asterisk-Users] Realtime Static/Dynamic Preference

[Douglas Garstang]

Actually, who says this is supposed to work anyways? When Asterisk fails to 
connect to the database when querying a number, does it have the logic to then 
fail over and try the same number in contexts that follow? If it doesn't, then 
there's no point.

-Original Message-
From: Douglas Garstang 
Sent: Tuesday, December 27, 2005 1:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Realtime Static/Dynamic Preference


Yes, I tied something like this. I included the database context first (the one 
that has the realtime switch) followed by the context that has the extensions 
locally. I shut the database down and Asterisk returns fast busy when dialling 
the number. Doesn't appear to work.

[OffNet]
#include "inc/OffNet/master.conf"

[OnNetFlat]
exten => 3250071,1,Dial(SIP/a00090101,20,tr)
exten => 3250072,1,Dial(SIP/a00090201,20,tr)
exten => 3250073,1,Dial(SIP/a00090301,20,tr)

[OnNetDB]
switch => Realtime/[EMAIL PROTECTED]

[OffNet]
#include "inc/OffNet/master.conf"

[Master]
include => OnNetDB
include => OnNetFlat

;
; User enters here.
;
[c_a00090101]
include => a00090101
include => Company1
include => Master
include => OffNet




-Original Message-
From: Kevin P. Fleming [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 27, 2005 10:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Realtime Static/Dynamic Preference


Douglas Garstang wrote:
> It seems that Asterisk gives priority to extensions in the extensions.conf 
> file over what's access in the db via the switch statement. For example, if 
> you have an entry in extensions.conf and realtime for the same extension, 
> Asterisk won't look in the db. 

This is true in general for Asterisk dialplans, it has nothing to do 
with Realtime.

Extensions defined in the context itself are always searched before any 
included contexts or switches. If you want to control the search order, 
you must put _all_ your extensions into separate contexts (by type or 
whatever other grouping you wish) and then use a 'master' context with 
include/switch statements in the order you wish them to be processed.
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RE: [Asterisk-Users] Realtime Static/Dynamic Preference

[Douglas Garstang]

Yes, I tied something like this. I included the database context first (the one 
that has the realtime switch) followed by the context that has the extensions 
locally. I shut the database down and Asterisk returns fast busy when dialling 
the number. Doesn't appear to work.

[OffNet]
#include "inc/OffNet/master.conf"

[OnNetFlat]
exten => 3250071,1,Dial(SIP/a00090101,20,tr)
exten => 3250072,1,Dial(SIP/a00090201,20,tr)
exten => 3250073,1,Dial(SIP/a00090301,20,tr)

[OnNetDB]
switch => Realtime/[EMAIL PROTECTED]

[OffNet]
#include "inc/OffNet/master.conf"

[Master]
include => OnNetDB
include => OnNetFlat

;
; User enters here.
;
[c_a00090101]
include => a00090101
include => Company1
include => Master
include => OffNet




-Original Message-
From: Kevin P. Fleming [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 27, 2005 10:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Realtime Static/Dynamic Preference


Douglas Garstang wrote:
> It seems that Asterisk gives priority to extensions in the extensions.conf 
> file over what's access in the db via the switch statement. For example, if 
> you have an entry in extensions.conf and realtime for the same extension, 
> Asterisk won't look in the db. 

This is true in general for Asterisk dialplans, it has nothing to do 
with Realtime.

Extensions defined in the context itself are always searched before any 
included contexts or switches. If you want to control the search order, 
you must put _all_ your extensions into separate contexts (by type or 
whatever other grouping you wish) and then use a 'master' context with 
include/switch statements in the order you wish them to be processed.
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Re: [Asterisk-Users] Asterisk does not handle call from a Cisco IAD correctly

[James Sizemore]

I think I found what is munging up the peer lookup:

This call from another Asterisk box starts:

<-- SIP read from 192.168.69.254:5060:

The peer lookup that fail reads:

<-- SIP read from 192.168.7.250:52141:

Asterisk seem to be thrown off by the fact that the return port is not
5060, and fails the peer lookup.  This is a * bug then. I have 
documented it with both 1.0.9 and 1.2.1. Time to dig through the sip code.



James Sizemore wrote:
when my Cisco IAD send a call to my Asterisk gateway the gateway treats 
it as if I don't have a peer statement in sip.conf, when I do. Here are 
the first two packets, notice the "Found no matching peer or user for 
'192.168.7.250:50437'" on the second packet. Any one seen this before, 
or have a clue as to the problem?  Asterisk 1.0.9


sip.conf:
[bna-vonx-iad]
type=friend
context=trusted-out
host=192.168.7.250
canreinvite=no


Sip read:
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP  192.168.7.250:5060;branch=z9hG4bK1A60
From: "James Sizemore" ;tag=19D8A640-5E9
To: 
Date: Wed, 06 Mar 2002 00:27:08 GMT
Call-ID: [EMAIL PROTECTED]
Supported: 100rel,timer
Min-SE:  1800
Cisco-Guid: 3128236623-802099670-2154346748-2004044536
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, 
SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER

CSeq: 101 INVITE
Max-Forwards: 6
Remote-Party-ID: "James Sizemore" 
;party=calling;screen=yes;privacy=off

Timestamp: 1015374428
Contact: 
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 191

v=0
o=CiscoSystemsSIP-GW-UserAgent 6047 8216 IN IP4 192.168.7.250
s=SIP Call
c=IN IP4 192.168.7.250
t=0 0
m=audio 16434 RTP/AVP 0
c=IN IP4 192.168.7.250
a=rtpmap:0 PCMU/8000
a=ptime:20

20 headers, 9 lines
Using latest request as basis request
Sending to 192.168.7.250 : 5060 (non-NAT)
Found no matching peer or user for '192.168.7.250:50437'
Found RTP audio format 0
Peer audio RTP is at port 192.168.7.250:16434
Found description format PCMU
Capabilities: us - 0x78e (gsm|ulaw|alaw|lpc10|g729|speex|ilbc), peer - 
audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined 
- 0x0 (nothing)

Looking for 615917 in default
list_route: hop: 
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.7.250:5060;branch=z9hG4bK1A60
From: "James Sizemore" ;tag=19D8A640-5E9
To: ;tag=as43478a8a
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
User-Agent: Memphis ISDN-NET PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0


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RE : [Asterisk-Users] TDM2400

[f6hqz-m]

Hello folks !

TDM2400 with "E" for echocan module is ok for me, replacing my old passive
cards.
No more echo issues now. I had many before to switch to this wonderfull card
!
Perfect for my use...

Here is an Asterisk SVN-branch-1.2-r7608M, in an old PII-400 MHz
Linux version 2.6.12-1-686 (gcc version 4.0.2 20050917 (prerelease) (Debian
4.0.1-8) "Etch".

My opinion : buy it WITH the echocan option, and don't forget to buy a
Centronics 50 pins mâle connector (not provided).

Best Regards,
Francois BERGERET,
France.


-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Guillermo
Salas M
Envoyé : jeudi 22 décembre 2005 17:01
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : [Asterisk-Users] TDM2400


Hi all, I was checking the TDM2400 features and seems to me very
interesating. I think is that I need :)

I want to know your experience with this card and if you know abouts bugs,
configuration and everithing thah I need to know before acquire it :)

The http://www.voipsupply.com/product_info.php?products_id=1115 is necesary
?

Best regards,

-- 
Guillermo Salas M.
Telconet S.A. Manta
Calle 15 y Av. 24 Esq.
Phone : 593 5 262 8071
Mobile: 593 9 985 5138
SIP   : [EMAIL PROTECTED]
e-mail: [EMAIL PROTECTED]
www   : http://www.telconet.net
http://www.telcocarrier.net

Linux User: 255902
Soporte en Linea en http://www.manta.telconet.net

Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html

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[Asterisk-Users] MSN Messenger / Windows messenger Passport service With asterisk any one ?

[Rehan Ahmed]

Hello,
 
Has any one been able to recveive a call from asterisk to msn ( not windows messenger by registering on asterisk) but on regular as hotmail id.
 
Please contact me even if there is a charge for it.
 
Rehan
 
-- Rehan Ahmed AllahWalahttp://www.SuperTec.com - Tommrow's Technology, Today.http://www.didx.net - DID Number Exchange and Peering Service.

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[Asterisk-Users] polycom sip slower than grandstream

[Dean Collins]

I have a polycom 501, for some reason asterisk always shows
the round trip time to it as being significantly higher than the 2
grandstreams, all 3 are on the same lan.

 

 

 

Grandstream 40/40    192.168.16.40    D 
255.255.255.255  5060 OK (4 ms)

Grandstream 31/31    192.168.16.31    D 
255.255.255.255  5060 OK (4 ms)

Polycom   30/30    192.168.16.30    D 
255.255.255.255  5060 OK (78 ms)

 

 

 

Any thoughts?

 

 

Dean






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RE: [Asterisk-Users] Blackberry SIM card

[Alexander Lopez]

You do not need the BES server.. It is nice for total wireless
syncronization but not need for it to work. 

The BB will work in three ways:

BES server, Married with Exchange server or Lotus notes. 

Internet only, you are given an address like
[EMAIL PROTECTED], you them forward your emails to it, you
can change the way email get sent so it looks like your address in stead
of [EMAIL PROTECTED] (does not work to well with Domain Keys or
SPF)

Redirector, SW is loaded on your machine (PC) and interacts with
Outlook, all mail is sent to your mailbox in an encripted file and then
the redirector 'redirects it' out to the user via your PC. 


As far as Asterisk support goes, the only support that the BlackBerry
has for Asterisk would be the Bluetooth interface.

Other nice features that are on but not limited to the BB are:

Fax to your email as a PDF. (to read you either need a space lens from
the Hubbel Telescope, or a space Electron Microscope!)

Your VM left on asterisk include caller information so that returning
calls is as easy as select and click.


As far as a SIM card, if your unit is 'unlocked' you can put in any GSM
Sim card and activate service with whiomever you fancy, provided they
support BlackBerry.
.
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Kerry Garrison
> Sent: Tuesday, December 27, 2005 1:06 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Blackberry SIM card
> 
> To get a sim card you need service from T-Mobile. Any cell 
> phone, land line, sat phone, etc will work with Asterisk 
> depending on what you mean "work with *". You can set up an 
> phone number as a custom extension. If you mean you want to 
> setup the blackberry as a sip phone that communicates with 
> your Asterisk server, I don't really know how you use it 
> within the Blackberry but if you have the Blackberry 
> Enterprise Service you can send out a setting to the 
> Blackberry to configure a SIP server. To make this all happen 
> you will need:
> 
> Blackberry (you already have that)
> SIM Card (requires T-Mobile service account) Windows 
> 2000/2003 Server (maybe you have one, maybe not) Blackberry 
> Enterprise Server (about $1,500 for a 5 user license)
> 
> This is based on my LIMITED experience servicing a client 
> that has a few Blackberrys, I have never touched the SIP 
> settings on theirs, I have only seen them in the BES setup. 
> If there is something different that I don't know about, 
> please let me know.
> 
> Kerry Garrison
> Publisher - http://GeekGazette.com - http://VOIPSpeak.net
> (949) 502-7819 x200 - [EMAIL PROTECTED] 
> http://www.techdatapros.com 
>  
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Robert Rawlinson
> Sent: Tuesday, December 27, 2005 9:50 AM
> To: Asterisk
> Subject: [Asterisk-Users] Blackberry SIM card
> 
> I acquired a Blackberry 7100T over Christmas. I had heard it 
> will work with
> * and that is what I want to do with it. But I think it needs 
> a SIM card to make it work. If this is true how do I go about 
> getting a SIM card for it and how to set it up? Thanks for 
> any help you can offer.
> Bob Rawlinson
> 
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RE: [Asterisk-Users] Asterisk lines go into PBX?

[Michael Collins]

Doug,

 

You might also check out the
wiki.  There is a great deal of information regarding the connection of
Asterisk to “legacy” systems.  I wrote the one on connecting
to an NEC NEAX 2400.  Here’s the main wiki page:

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

Here’s the page for
connecting to other PBX’s:

http://www.voip-info.org/wiki/view/Asterisk+legacy+integration

 

Hopefully your PBX is
listed.  Some things to consider when connecting Asterisk to a PBX:

Analog or digital?

Will Asterisk be “line
side” or “station side”?  In other words, from the point
of view of the PBX, will the connection(s) coming from Asterisk be CO lines or
PBX extensions?

 

Each choice has its
advantages and disadvantages.  The Asterisk community will be happy to
help you answer your questions.  Would you mind posting some more
information on what you’d like to do with Asterisk and your existing
system?  Also, could you tell us which PBX you are using (Nortel, Avaya,
NEC, Panasonic, etc.) and what kind of interfaces card(s) you have?  I’m
an NEC guy but there are lots of other PBX guys for all sorts of PBXs out
there.

 

Enjoy Asterisk!  It’s
a whole new world, let me tell you…

 

-MC

 

 

 

 

Date: Mon, 26 Dec 2005
15:34:22 -0500

From: "Dean
Collins" <[EMAIL PROTECTED]>

Subject: RE:
[Asterisk-Users] Asterisk lines go into PBX?

To: "Asterisk Users
Mailing List - Non-Commercial Discussion"

  

Message-ID:

  <[EMAIL PROTECTED]>

Content-Type: text/plain; charset="us-ascii"

 

> Hi Doug,

> There are a number of
hardware providers for either pstn or isdn

> interfaces (co lines).
Why not check out www.digium.com

> 

> A good way to get
started is by using

> http://asteriskathome.sourceforge.net/

> Which is an automated
installation and configuration program.

> 

> Welcome to Asterisk.

> 

> Cheers,

> Dean

 

 

> > -Original
Message-

> > From:
[EMAIL PROTECTED] [mailto:asterisk-users-

> >
[EMAIL PROTECTED] On Behalf Of Doug

> > Sent: Monday,
December 26, 2005 1:58 PM

> > To: asterisk-users@lists.digium.com

> > Subject:
[Asterisk-Users] Asterisk lines go into PBX?

> > 

> > How can Asterisk
lines be configured like

> > central office
lines feeding into a PBX?

> > 

> > Has anyone done
this before?

> > 

> > What about
"rollover" / hunt groups if

> > a line is busy?

> > 

> >
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> >    http://lists.digium.com/mailman/listinfo/asterisk-users

 






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RE: [Asterisk-Users] CDR_CSV stops writing, help!

[Tyler]

I wish I could on this box.  It is in the plans, but we can't do that
just yet.  Is this a known issue with the cdr_csv module in the 1.0
branch ??

tf.

On Tue, 2005-12-27 at 10:25, Alexander Lopez wrote:
> Tyler, 
> 
>   Can you upgrade to 1.2???
> 
> Alex 
> 
> > -Original Message-
> > From: [EMAIL PROTECTED] 
> > [mailto:[EMAIL PROTECTED] On Behalf Of Tyler
> > Sent: Tuesday, December 27, 2005 9:40 AM
> > To: asterisk-users@lists.digium.com
> > Subject: [Asterisk-Users] CDR_CSV stops writing, help!
> > 
> > Hi list..
> > 
> > Using asterisk 1.0.10 and the cdr_mysql addon to write CDR 
> > records to a MySQL table.  That part works great.  The issue 
> > is that I also need the Master.csv text CDR log and thusly 
> > have the cdr_csv.so module loaded. 
> > The problem is, after 10-15 mins of activity, it just.. stops 
> > writing. 
> > tail -f Master.csv no longer shows anything, restarting 
> > asterisk doesn't flush any buffers, etc.  It just stops.
> > 
> > Issuing a reload or reloading the cdr_csv.so module gets it 
> > working again for another few minutes, but I don't get why 
> > it's stopping.
> > 
> > I've got plenty of disk space available and the Master.csv 
> > file is only 500k in size (right now).
> > 
> > Anyone come across this before?
> > 
> > tf.
> > 
> > 
> > 
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Re: [Asterisk-Users] SIP ENUM Daemon

[Klaus Darilion]

Hi Nahid!

Why do you want to do this? What about the ENUM resolvers inside 
asterisk? There is the old EnumLookup application, and the new the 
ENUMLOOKUP function (with plenty of features). Have you tried them?


If you do not want to use asterisk's internal ENUM resolvers, you could 
also use (open)ser just for you purpose. It makes the ENUM lookup and 
replies with 302.


regards
klaus

Nahid Hossain wrote:

Hello,

I am trying to develop a simple but fast application/daemon to take SIP 
invites, convert them into ENUM queries, send those queries to an ENUM 
server (likely residing on the same hardware as the daemon), get back an 
ENUM response and convert that to a SIP 302 (or other 300 level) 
redirect response.


 

Before developing this type of daemon, I just want to make sure about 
the expected daemon related to performance. If anyone has already done 
this type of work, please give some comments regarding the following 
performance issues, whether the following challenges are possible or not 
in real scenario.


 


   1. Daemon responds to an ENUM query in 0.7 milliseconds and answer
  16,000 queries per second.
   2. Daemon should add less than 1 millisecond to that roundtrip so a
  SIP invite should get a 300 redirect within 1.7 millisecond.

 

I would appreciate if anyone can help me with relevant information. If 
the above challenges are not possible in real scenario, then appreciate 
for any near numeric figure which are possible to maintain in daemon.


 


Thanks

Nahid

 





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Re: [Asterisk-Users] PRI outgoing caller ID stopped working

[Andrew Kohlsmith]

On Saturday 24 December 2005 16:40, Kevin P. Fleming wrote:
> Interestingly, some systems I manage also began exhibiting this behavior
> in the past ten days or so. I have been working with the telco and they
> too show the Calling Number being received as expected over the PRI, but
> yet the far end receives 'Unknown' or 'Out of Area' depending on their
> CLID display device.
>
> I will continue to try to debug it, but I can't back down the code on
> that box to an older version for comparison of the PRI traffic; if you
> can do so, that would be most helpful.

rev 5552 of asterisk, rev 208 of libpri, rev 877 (current) of zaptel... I 
still have this problem.  Now this code is significantly older (with the 
exception of zaptel) than what I was running about a month ago when it was 
known to work...

No great news yet, but at least it's a datapoint.  :-(

-A.
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Re: [Asterisk-Users] Cisco dtmf

[Greg Oliver]

I use:

# Enable_VAD (1-enabled, 0-disabled)
enable_vad: "0"
dtmf_inband: "1"
dtmf_outofband: "never"
dtmf_avt_payload: "101"

and it works well for me.  Sometimes going through a callmanager I have
to set outofband to avt to get dialtone sent though.

On Tue, 2005-12-27 at 16:05 +0100, Tomislav Parcina wrote:
> I'm trying to set up call transfer and automon options. They work fine 
> with ZAP lines (analog telephone) and with Grandstream Budgetone 102. I 
> have problem with Cisco 7905 and 7940. I think that problem is with dtmf 
> signalization. 
> 
> This is my configuration in 7940 
> dtmf_inband: 1
> dtmf_outofband: none  
> dtmf_db_level: 3
> 
> And 7905
> AudioMode:0x
> 
> Is my configuration wrong or it doesn't work with Cisco phones?
> 
> Thank you for your time!
> 
> 

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[Asterisk-Users] UK, Disconnect supervision

[Peter Hoppe]

Hello!

This is actually less a question than some information, if anyone else 
struggles with the same issue.


I am located in the UK and use a Sipura-3000 adapter to connect to a BT 
line (via fxo port). One problem I had was that disconnect supervision 
didn't work:


Some caller phones me (my adapter)
adapter goes off-hook (answers call)
caller hangs up
adapter doesn't realize and stays off hook.

I researched into it and found that the BT exchange delivers a CPC (0.1 
sec) upon the caller's hangup and a disconnect tone about 3 seconds 
later. The tone lasts 6 seconds and remains constant during that time. I 
recorded the disconnect tone with Cool Edit and did a frequency analysis 
on it and got the following components:


400Hz/-56dB +
1200Hz/-69dB +
2000Hz/-65dB +
2800Hz/-59dB +
3600Hz/-59dB

If normalized with -56 dB reference level, I get

400Hz/0dB +
1200Hz/-13dB +
2000Hz/-9dB +
2800Hz/-3dB +
3600Hz/-3dB

I suspect that the harmonics (1200, 2000, 2800, 3600)Hz may not have 
come from the Exchange, but were distortions.


After the 6 seconds the tone stopped and there was succession of two 
'clicks' on the line (another CPC? haven't looked into that); both were 
0.3 secs apart.


I tried to use both, the CPC and the disconnect tone in the Sipura-3000 
settings. Unfortunately I couldn't get disconnect supervision via the 
disconnect tone to work. In a post on the voxilla forum


http://voxilla.com/PNphpBB2-viewtopic-t-2904.html

I found some further info on disconnect tone in the UK - unfortunately I 
couldn't get those settings to work. I also looked into BT's SIN notes 
(on http://www.sinet.bt.com , notes 350, 351) but failed to see further 
information on what tone they exactly deliver for a disconnect event.


However, the CPC did work. I set the adapter to a CPC minimum time of 
0.05 seconds, and from then on it recognized remote disconnection in 
every test phone call. If I set the time too short I got problems - for 
incoming calls the adapter started to mistake the incoming ringing for 
disconnect events and simply wouldn't go off hook anymore.


Now, these are the settings I configured my adapter with:

Settings: 
spa-3000-setup-web-page/PSTN-line-tab/PSTN-disconnect-detection-section:


'Detect CPC' => 'yes'
'Min CPC Duration' => '0.05'

I also have
'Detect Disconnect Tone' => 'no' [as it didn't work]
'Disconnect Tone' => '[EMAIL PROTECTED],[EMAIL PROTECTED];5(5/5/1+2)'
'Detect Polarity Reversal' => 'Yes'
'Detect PSTN Long Silence' => 'yes'
'PSTN Long Silence Duration' => '150'
'Detect VoIP Long Silence' => 'no'
'VoIP Long Silence Duration' => '30'

but the relevant values are 'Detect CPC' => 'yes' and 'Min CPC Duration' 
=> '0.05'



I hope this helps, if anyone struggles with unrecognized disconnects.

God bless, Peter


CPC: Calling Party Control: A short break in the line current in the 
called party's phone line when the calling party hangs up.

(def. from http://www.vikingelectronics.com/glossary/telecom-term.php)
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Re: [Asterisk-Users] ISDN/CAPI outgoing calls - weirdness with ringing

[Armin Schindler]

On Sun, 25 Dec 2005, Michael J. Tubby G8TIC wrote:
> > On Sat, 24 Dec 2005, Michael J. Tubby G8TIC wrote:
> > > I changed the dial-string to include flags 'ob' as you mentioned
> > > (below)
> > > and now I get the following when I dial a BT phone number
> > > 
> > > - dial number, get:
> > > 
> > > Proceeding (in 100) briefly
> > > 
> > > - after a second or so:
> > > 
> > > Ringng Destination (in 180)
> > > 
> > > - double ringing tone:
> > > 
> > > BT style ringing generated by the exhange
> > > Cisco phone US-style ringing (generated by the phone)
> > > 
> > > these are overlaid on each other (mixed together)
> > > 
> > > 
> > > My hunch is that there's something not right with the call set up
> > > sequence
> > > and CAPI handling.
> > 
> > This is not a problem of CAPI. When you specify 'b' for early-b3, you
> > will
> > get the tones from the switch. If your phone adds its own tone, even when
> > it
> > receives progress tones, then it is incorrect (maybe wrong setup).
> > 
> > Armin
> > 
> 
> 
> However the difference that I see looking at the Cisco 7960 phone which
> shows a version of the SIP messages on its status line is:
> 
> 100 Proceeding
> 183 Session Progress
> 180 Ringng Destination
> 
> the order of which varies and depends on the dialled number.
> 
> Some dialled numbers go: 100->183->180 and these produce one set
> of alerting/ringing correctly.
> 
> Some dialled numbers go: 100->183 and stay in state 183 until the called
> party answers - these are the ones that produce no ringing.

Can you provide a verbose log level 5 with 'capi debug' ?
I would like to compare the capi messages. Maybe the switch just send an 
alerting message.
 
> If I add the 'o' to the existing 'b' flag then dial it appears to change the
> behaviour so that the phone goes 100-180 for all calls but some give
> me a single (phone generated US style ring) while others give the 'double
> ringing'.  The ones that produce double ringing are the ones that would
> have rung before, while the ones that now produce ringing (from the
> exchange) are the ones that used to be silent.

When using 'o', chan_capi is doing early-b3 from the beginning before 
sending any digits and you will get b3-data in each case.
Please send me a debug log of a connection with double ring-tone (no 183)
as well.
 

Armin
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Re: [Asterisk-Users] Realtime Static/Dynamic Preference

[C F]

Take a look at the following page (you might be able to change the priority):
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf+sorting

On 12/27/05, Douglas Garstang <[EMAIL PROTECTED]> wrote:
> It seems that Asterisk gives priority to extensions in the extensions.conf 
> file over what's access in the db via the switch statement. For example, if 
> you have an entry in extensions.conf and realtime for the same extension, 
> Asterisk won't look in the db.
>
> Anyone know if there's a way to switch this around, and have Asterisk look in 
> realtime first and then if it isn't accessible, use what's in 
> extensions.conf? Reason is for availability. If database is down, still allow 
> Asterisk to use what's in extensions.conf as a backup.
>
> Thanks,
> Doug.
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[Asterisk-Users] Asterisk does not handle call from a Cisco IAD correctly

[James Sizemore]

when my Cisco IAD send a call to my Asterisk gateway the gateway treats 
it as if I don't have a peer statement in sip.conf, when I do. Here are 
the first two packets, notice the "Found no matching peer or user for 
'192.168.7.250:50437'" on the second packet. Any one seen this before, 
or have a clue as to the problem?  Asterisk 1.0.9


sip.conf:
[bna-vonx-iad]
type=friend
context=trusted-out
host=192.168.7.250
canreinvite=no


Sip read:
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP  192.168.7.250:5060;branch=z9hG4bK1A60
From: "James Sizemore" ;tag=19D8A640-5E9
To: 
Date: Wed, 06 Mar 2002 00:27:08 GMT
Call-ID: [EMAIL PROTECTED]
Supported: 100rel,timer
Min-SE:  1800
Cisco-Guid: 3128236623-802099670-2154346748-2004044536
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, 
SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER

CSeq: 101 INVITE
Max-Forwards: 6
Remote-Party-ID: "James Sizemore" 
;party=calling;screen=yes;privacy=off

Timestamp: 1015374428
Contact: 
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 191

v=0
o=CiscoSystemsSIP-GW-UserAgent 6047 8216 IN IP4 192.168.7.250
s=SIP Call
c=IN IP4 192.168.7.250
t=0 0
m=audio 16434 RTP/AVP 0
c=IN IP4 192.168.7.250
a=rtpmap:0 PCMU/8000
a=ptime:20

20 headers, 9 lines
Using latest request as basis request
Sending to 192.168.7.250 : 5060 (non-NAT)
Found no matching peer or user for '192.168.7.250:50437'
Found RTP audio format 0
Peer audio RTP is at port 192.168.7.250:16434
Found description format PCMU
Capabilities: us - 0x78e (gsm|ulaw|alaw|lpc10|g729|speex|ilbc), peer - 
audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined 
- 0x0 (nothing)

Looking for 615917 in default
list_route: hop: 
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.7.250:5060;branch=z9hG4bK1A60
From: "James Sizemore" ;tag=19D8A640-5E9
To: ;tag=as43478a8a
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
User-Agent: Memphis ISDN-NET PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0 




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Re: [Asterisk-Users] asterisk AVM C2 again

[Armin Schindler]

On Fri, 23 Dec 2005, stéphane plichon wrote:
> Armin Schindler wrote:
> > 
> > 
> > Please create a verbose log of level 5 with 'capi debug'...
> > 
> > Armin
> > 
> > 
> debug for incoming call:
...
>> CAPI INFO 0x3302: Protocol error layer 2
...
>> CAPI INFO 0x3302: Protocol error layer 2
>   == AVM2: CAPI Hangingup
>   == AVM2: Interface cleanup PLCI=0x202

It looks like the call is signaled on both ports !?
And the error messages "Protocol error layer 2" show that you are using a
wrong protocol setting.

> -- Executing Goto("|Íç·|Íç·1/9881-1", "default|s|1") in new stack
> -- Goto (default,s,1)
> -- Executing Dial("<135>Dec 23 13:15:07 asterisk[7398]:
> VERBOSE[7398]: -- Goto (defa", "SIP/202&SIP/201&SIP/200|3
>  in new stack
> -- Called 202
> -- Called 201
> -- Called 200
>   == Starting 68.100.142:28479 at 005 12:15:07 GMT,CMU/8000,0 failed so
> falling back to exten 's'
>   == Starting 68.100.142:28479 at 005 12:15:07 GMT,s,0 still failed so
> falling back to context 'default'
> -- Executing Dial("68.100.142:28479",
> "SIP/202&SIP/201&SIP/200|30|tT") in new stack
> Dec 23 13:15:07 WARNING[7399]: pbx.c:445 pbx_exec: Stack overflow,
> cannot create another stack

I cannot tell anything about this one. Maybe someone else has an idea what pbx
is doing here.

Armin
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Re: [Asterisk-Users] Transfer

[Tobias Wolf]

Victor Alvarez schrieb:

Hi,
 I'm afraid I don't know how to use the command Transfer. 


I am also interested how the command "Transfer" should be used.

I am aware of the possibility to add the option t or T to dial, so #33 
transfers the call to extension 33.


Is there any use of this command in the dialplan? If I want to redirekt 
a call because of the choices of a caller goto() or dial() does the job.


Thx for any background :)

Have a happy new year, folks.

Tobias
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RE: [Asterisk-Users] Blackberry SIM card

[Kerry Garrison]

To get a sim card you need service from T-Mobile. Any cell phone, land line,
sat phone, etc will work with Asterisk depending on what you mean "work with
*". You can set up an phone number as a custom extension. If you mean you
want to setup the blackberry as a sip phone that communicates with your
Asterisk server, I don't really know how you use it within the Blackberry
but if you have the Blackberry Enterprise Service you can send out a setting
to the Blackberry to configure a SIP server. To make this all happen you
will need:

Blackberry (you already have that)
SIM Card (requires T-Mobile service account)
Windows 2000/2003 Server (maybe you have one, maybe not)
Blackberry Enterprise Server (about $1,500 for a 5 user license)

This is based on my LIMITED experience servicing a client that has a few
Blackberrys, I have never touched the SIP settings on theirs, I have only
seen them in the BES setup. If there is something different that I don't
know about, please let me know.

Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com 
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Rawlinson
Sent: Tuesday, December 27, 2005 9:50 AM
To: Asterisk
Subject: [Asterisk-Users] Blackberry SIM card

I acquired a Blackberry 7100T over Christmas. I had heard it will work with
* and that is what I want to do with it. But I think it needs a SIM card to
make it work. If this is true how do I go about getting a SIM card for it
and how to set it up? Thanks for any help you can offer.
Bob Rawlinson

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Re: [Asterisk-Users] Realtime Static/Dynamic Preference

[Kevin P. Fleming]

Douglas Garstang wrote:
It seems that Asterisk gives priority to extensions in the extensions.conf file over what's access in the db via the switch statement. For example, if you have an entry in extensions.conf and realtime for the same extension, Asterisk won't look in the db. 


This is true in general for Asterisk dialplans, it has nothing to do 
with Realtime.


Extensions defined in the context itself are always searched before any 
included contexts or switches. If you want to control the search order, 
you must put _all_ your extensions into separate contexts (by type or 
whatever other grouping you wish) and then use a 'master' context with 
include/switch statements in the order you wish them to be processed.

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[Asterisk-Users] Blackberry SIM card

[Robert Rawlinson]

I acquired a Blackberry 7100T over Christmas. I had heard it will work 
with * and that is what I want to do with it. But I think it needs a SIM 
card to make it work. If this is true how do I go about getting a SIM 
card for it and how to set it up? Thanks for any help you can offer.

Bob Rawlinson

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RE: [Asterisk-Users] Realtime Static/Dynamic Preference

[Douglas Garstang]

Thanks, but static isn't an option. Users will have the ability to make changes 
to their dialplan via a web portal. Doing a 'reload' every few seconds/minutes 
is even less viable especially when you consider that a reload deletes all the 
SIP subscriptions.

-Original Message-
From: BJ Weschke [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 27, 2005 10:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Realtime Static/Dynamic Preference


On 12/27/05, Douglas Garstang <[EMAIL PROTECTED]> wrote:
> It seems that Asterisk gives priority to extensions in the extensions.conf 
> file over what's access in the db via the switch statement. For example, if 
> you have an entry in extensions.conf and realtime for the same extension, 
> Asterisk won't look in the db.
>
> Anyone know if there's a way to switch this around, and have Asterisk look in 
> realtime first and then if it isn't accessible, use what's in 
> extensions.conf? Reason is for availability. If database is down, still allow 
> Asterisk to use what's in extensions.conf as a backup.
>

 If it can't be done via config, it's probably not too hard to change
the code to do it. Be careful with switch though as that is the most
demanding (query wise) of all the realtime engines at the moment. It
will query the database with every call coming through the dial plan.
If you know when your DP is changing, you may opt to do static instead
and just reload from DB when your DP changes.

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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Re: [Asterisk-Users] Polycom IP301 time changing

[Mojo with Horan & Company, LLC]

By config file do you by chance happen to mean dhcpd.conf?  I'm pretty 
sure there are settings like this in ipmid.cfg or sip.cfg, but it could 
be that the dhcpd.conf one is conflicting.  In mine I have


option time-offset -32400;

Again, I'm not totally sure there is a time offset in the phone or its 
config files, but I'm assuming there is for the purposes of this 
suggestion ;)  And, if for example you set something with sip.cfg or 
ipmid.cfg, but subsequently adjust a setting on the phone that 
conflicts, I think that's when you get a -phone.cfg -- so 
check that one for time offsets too :)


Moj



Jonathan k. Creasy wrote:

I have 13 Polycom IP301's where the clock keeps resetting to a +5
offset. I can change the config file to show -5, change it to -5 on the
phone and after an hour or so the phone will update itself back to +5. 


Anyone have any ideas? The other 70+ phones are not exhibiting this
behavior. 


-Jonathan
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--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
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Re: [Asterisk-Users] Realtime Static/Dynamic Preference

[BJ Weschke]

On 12/27/05, Douglas Garstang <[EMAIL PROTECTED]> wrote:
> It seems that Asterisk gives priority to extensions in the extensions.conf 
> file over what's access in the db via the switch statement. For example, if 
> you have an entry in extensions.conf and realtime for the same extension, 
> Asterisk won't look in the db.
>
> Anyone know if there's a way to switch this around, and have Asterisk look in 
> realtime first and then if it isn't accessible, use what's in 
> extensions.conf? Reason is for availability. If database is down, still allow 
> Asterisk to use what's in extensions.conf as a backup.
>

 If it can't be done via config, it's probably not too hard to change
the code to do it. Be careful with switch though as that is the most
demanding (query wise) of all the realtime engines at the moment. It
will query the database with every call coming through the dial plan.
If you know when your DP is changing, you may opt to do static instead
and just reload from DB when your DP changes.

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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Re: [Asterisk-Users] Login incorrect on ZIP2 phones when checking voicemail

[John Novack]

Dan Elder wrote:


Hi all, I have rolled out a few Zultys ZIP2 phones, and they seem to work fine, 
except when trying to check voicemail. If we go into comedian mail, we are 
prompted for a extension #, then a password. The ext # transmits properly, but 
the password is not being heard by asterisk. The CLI output says the password 
entered was blank, but it wasn't.. any idea what could cause this? I've been 
toying w/the dialplan, but nothing I've done has worked, is there something 
else I need to setup on these pohones? I have a
Linksys SAP841 which transmits the password info properly, it's just the ZIP2s 
that aren't sending the pass.
 


Have you found a way to get the  MWI to work on these phones??

Personally I don't think much of the ZIP2 ,  overpriced, lack of any 
speakerphone or display, no PoE and no wall mounting ability.


John Novack



Thanks in advance & happy new year!

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[Asterisk-Users] Realtime Static/Dynamic Preference

[Douglas Garstang]

It seems that Asterisk gives priority to extensions in the extensions.conf file 
over what's access in the db via the switch statement. For example, if you have 
an entry in extensions.conf and realtime for the same extension, Asterisk won't 
look in the db. 

Anyone know if there's a way to switch this around, and have Asterisk look in 
realtime first and then if it isn't accessible, use what's in extensions.conf? 
Reason is for availability. If database is down, still allow Asterisk to use 
what's in extensions.conf as a backup.

Thanks,
Doug.
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[Asterisk-Users] Cisco 7912G through NAT, problems with tones detection.

[Diego Mariano Velo]

Hi, i have a cisco 7912G with SIP firmware, its connect to the asterisk
through nat. The only problems is in the voice mailasterisk not
detect the tones, therefore i cant access to my voice mail extension.

Thanks in advance.

 Diego.
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[Asterisk-Users] Login incorrect on ZIP2 phones when checking voicemail

[Dan Elder]

Hi all, I have rolled out a few Zultys ZIP2 phones, and they seem to work
fine, except when trying to check voicemail. If we go into comedian mail, we
are prompted for a extension #, then a password. The ext # transmits
properly, but the password is not being heard by asterisk. The CLI output
says the password entered was blank, but it wasn't.. any idea what could
cause this? I've been toying w/the dialplan, but nothing I've done has
worked, is there something else I need to setup on these pohones? I have a
Linksys SAP841 which transmits the password info properly, it's just the
ZIP2s that aren't sending the pass.

Thanks in advance & happy new year!

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RE: [Asterisk-Users] LD_LIBRARY_PATH

[Douglas Garstang]

Add it 
to /etc/ld.so.conf

  -Original Message-From: Kanishka Somaratne 
  [mailto:[EMAIL PROTECTED]Sent: Monday, December 26, 2005 9:51 
  PMTo: asterisk-users@lists.digium.comSubject: 
  [Asterisk-Users] LD_LIBRARY_PATH 
  HiI set the LD_LIBRARY_PATH and when i reboot i have to set it again. 
  how do i set it in linux to load it when the server 
  reboots.RegardsKani 
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Re: [Asterisk-Users] channel monitoring whisper mode?

[Saul Diaz]

Script Head wrote:

As this isn't a part of *, has anyone accompilished a whisper mode in 
yet? What I am looking for is an ability for to say something while 
monitoring a channel and the agent being able to hear what I say while 
the called party is not.


ScriptHead



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I am thinking to develop one.

regards
saul
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RE: [Asterisk-Users] Stay away from Grandstream!

[The VoIP Connection]

And buy your phones from a reputable dealer who will provide you with
support.  Grandstream's policy (and sipura, snom, polycom, etc.) is to
provide warrantee service through their resellers. We have never had them
reject a properly documented RMA. -Mike

Michael Crown
Managing Partner
www.thevoipconnection.com
321.989.6728 ext. 611
sip:[EMAIL PROTECTED]

> -Original Message-
> From: Chris Albertson [mailto:[EMAIL PROTECTED] 
> Sent: Monday, December 26, 2005 11:01 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Stay away from Grandstream!
> 
> Maybe a better way to say it is "Know the limitations of the 
> GS phones and don't try and use them outside of those 
> limits."  Don't buy ANY phone you've not tested and used 
> yourself for use by a client.
> My GS phone has worked fine for years.  Even if it were to 
> fail and had to be replaced buying two is still cheaper then 
> one of some of the others.  The trick is to use them (or 
> anyhting else) only when you know it will work.  That said, 
> the GS 100 is not the best thing to put on a receptionist's desk.  
> 
> I've actually had pretty good luck, even getting to 
> exchangeemail one of thier engineers.
> 
> 
> --- Elene Kinsky <[EMAIL PROTECTED]> wrote:
> 
> > We have 2 GXP-2000 dead during automatic firmware upgrade. 
> Devices now 
> > send out only one ARP packet for default gateway resolution during 
> > boot and nothing more!
> > We've contact Grandstream support, but they cannot help. 
> Now we want 
> > to send devices to Grandstream for repair but they on longer reply 
> > mail!
> > GXP-2000 was very buggy on attended call transfer, and the problem 
> > resolved only after upgrading using latest firmware. Overall GXP is 
> > OK, but customer support is terrible. Stay away from them!
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> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> 
> 
> Chris Albertson
>   Home:   310-376-1029  [EMAIL PROTECTED]
>   Cell:   310-990-7550
>   Office: 310-336-5189  [EMAIL PROTECTED]
>   KG6OMK
> 
> 
>   
>   
> __
> Yahoo! for Good - Make a difference this year. 
> http://brand.yahoo.com/cybergivingweek2005/
> 
> 

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[Asterisk-Users] Strange IAX messages on the console

[Joseph Rothstein]

When I reload IAX, I get the following messages on the console:

asterisk_test*CLI> iax2 reload
  == Parsing '/etc/asterisk/iax.conf': Found
Dec 27 16:56:28 NOTICE[23015]: chan_iax2.c:8618 set_config: Ignoring
bindport on reload
Dec 27 16:56:28 NOTICE[23015]: chan_iax2.c:8658 set_config: Ignoring
bindaddr on reload
-- doing lookup for '172.16.10.2'
  == Loaded firmware 'iaxy.bin'
  == Parsing '/etc/asterisk/iaxprov.conf': Found
-- Loaded provisioning template 'default'
asterisk_test*CLI>

This does not seem to effect IAX as it is up and running, but would like to
get rid of these messages, or at least know why they are being generated.

My iax.conf file has bindport=4569, and bindaddr=0.0.0.0

Regards to all,
Joe


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Re: [Asterisk-Users] spandsp & fax

[Dov Bigio]

Hi BJ, Kristof,

It worked!

I am using the version at
http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.2pre21c/asterisk-1.2.x/.

I think I had bad symlinks on /usr/local/lib and by reading the tutorial on
AsteriskGuru I found that... (The previously installed version of spandsp
has been 0.0.3, but now you have installed version 0.0.2. The problem is
that the installation of version 0.0.3 creates a symlink, which is not
replaced by installation of version 0.0.2. So the symlink points to the
library of version 0.0.3, which actually does not exist.). I simply deleted
all files related to spandsp from this directory and installed it again!

Thank you
Dov


- Original Message - 
From: "Kristof Hardy" <[EMAIL PROTECTED]>
To: "Dov Bigio" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List -
Non-CommercialDiscussion" 
Sent: Tuesday, December 27, 2005 12:59 PM
Subject: Re: [Asterisk-Users] spandsp & fax


> Dov Bigio wrote:
> > I am using Asterisk 1.2.1 and followed instructions on
> > http://www.asteriskguru.com/tutorials/spandsp.html to install faxing
> > capability on my server.
>
> what platform are you running on? (wich distro?)
> Does the make of the app_txfax and app_rxfax work out well?
>
>
>
>


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Re: [Asterisk-Users] "one touch record" on asterisk 1.2.1 uses monitor and not mixmonitor

[BJ Weschke]

On 12/27/05, Vikas <[EMAIL PROTECTED]> wrote:
> How to make "one touch record" on asterisk 1.2.1 use mixmonitor app ?
>
> In res_features.c line line 469:
> monitor_app = pbx_findapp("Monitor")
>
> How to make pbx_findapp return mixmonitor ?
>

 You'd need to make a few more changes to res_features.c than just the
pbx_findapp line you've pointed out. I suspect that this and the agent
based recording will get cut over to MixMonitor in the next major
release version. MixMonitor itself is still relatively new.

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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[Asterisk-Users] Callerid ID lookup program updated (CID_rewrite v1.2)

[Technical Support]

Many of you current 
cid_rewrite (v1.0.0) users probably noticed that your 411 lookup is broken, 
thanks to another change by the 411.com folks.  So we fixed it :)  The 
latest changes include:
1. Adapt to new 
411.com format
2. Improved address 
conversion and extraction from reverse lookup (removal of odd characters, 
addition of commas)
3. Addition of 
"action" variable in callerid database to allow you to build easy call 
screening
4. General code 
cleanup
5. Check the UPGRADE 
section of the README for more info on how to update
 
For those of you not 
familiar with the cid_rewrite tool, here's the (original) feature 
summary:
1. 
Standardize incoming caller-id numbers to adhere to US dialing code; 
NANPA numbers are reformatted to 1+10, international numbers become 
011 (this is customizable with a little PHP 
knowledge).
2. Look 
up the associated caller-id name in a mysql table.
3. If 
not found in the DB, it attempts a reverse-lookup on 411.com (and extracts the 
City name even if the number was not found).
4. If 
not found on 411.com, it attempts a reverse-lookup on Google.com; if not found, 
it falls back to pulling the rate-center from 
telcodata.us.
5. If 
available, the address associated with the phone-number is also extracted and 
inserted into the DB.
 
The latest script is 
available for download from www.generationd.com.  We now have the 
full package available for download.  Note that we are picking up 
development as the original author has updated the script in a long 
time.
 
Regards,
Michelle
 
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Re: [Asterisk-Users] newbie question about making outbound call

[Moises Silva]

Hi Jason. It seems your doing things "right" whatever that means. I
think the problem is more hardware related. Sure you have line in the
FXO?? have you tried dialing directly from some IP Phone?? I have
several applications that relay on automatic call generation with
Asterisk Manager and a PHP classes i have. But, as i said, i think the
problem is related to the configuration of the card. what does ztcfg
-vv says? what does zttool says??

best regardsOn 12/25/05, Jason D. Wolfe <[EMAIL PROTECTED]> wrote:
Hello,Somehow I've missed something here, so hopefully I'll be able to provideenough of my setup to get some help.  I feel I'm very close to gettingit, but missing something none the less...1. I have a digium TDM400 with (2) FXO modules on channel 3 and 4 hooked
to two POTS lines.2. I have the following entry in zapata.conf file:usecallerid=yeshidecallerid=nocallwaiting=nothreewaycalling=yestransfer=yesechocancel=yesechotraining=yescallprogress=no
context=incomingsignalling=fxs_kschannel=>43. I have the following entry in extensions.conf[callAgent]exten=>outbound,1,Dial(Zap/4/phonenumber)   ;where phonenumber is a 10digit number
exten=>outbound,n,Playback(access-code) ; just for the sake of doingsomething!4. I am using Asterisk Java Manager AGI OriginateAction with thefollowing code in a jsp page running on a  tomcat server:
//manageAGIManagerConnection managerConnection;ManagerConnectionFactory factory;OriginateAction originateAction;ManagerResponse originateResponse;factory = new ManagerConnectionFactory();
managerConnection = factory.getManagerConnection("192.168.1.4","jason","nosaj111");  // connect to Asterisk and log inmanagerConnection.login
();originateAction = new OriginateAction();originateAction.setAsync(true);originateAction.setChannel("Zap/4");originateAction.setContext("callAgent");
originateAction.setExten("outbound");originateAction.setPriority(new Integer(1));originateAction.setTimeout(3000);originateResponse =managerConnection.sendAction
(originateAction, 3);6. when I execute the jsp page, I watch the console started with/usr/sbin/asterisk -cvvand I get the following message (I substituted phonenumber in for the 10digit number again)
*CLI>   == Parsing '/etc/asterisk/manager.conf': Found  == Manager 'jason' logged on from 192.168.1.3   > Channel Zap/4-1 was answered.-- Executing Dial("Zap/4-1", "Zap/4/phonenumber") in new stack
Dec 25 10:55:40 NOTICE[3989]: app_dial.c:1010 dial_exec_full: Unable tocreate channel of type 'Zap' (cause 0 - Unknown)  == Everyone is busy/congested at this time (1:0/0/1)-- Executing Playback("Zap/4-1", "access-code") in new stack
-- Playing 'access-code' (language 'en')  == Manager 'jason' logged off from 192.168.1.3  == Auto fallthrough, channel 'Zap/4-1' status is 'CHANUNAVAIL'-- Hungup 'Zap/4-1'
exten => outbound,1,Hangup()What I eventually want to accomplish is the following:I want a web user (using a JSP page I think) to be able to click abutton and cause asterisk to dial outbound on both FXO ports, wait for
an answer, play some files, accept some input, and bridge the two callstogether.am I on the wrong track?  is there anything that is standing out that Iam just not understanding here?  ANY comments will be much appreciated.
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[Asterisk-Users] "one touch record" on asterisk 1.2.1 uses monitor and not mixmonitor

[Vikas]

How to make "one touch record" on asterisk 1.2.1 use mixmonitor app ?

In res_features.c line line 469:
monitor_app = pbx_findapp("Monitor")

How to make pbx_findapp return mixmonitor ?

T
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[Asterisk-Users] TDD/TTY - How does one use this?

[Don Fanning]

I'm trying to look for documentation on how the TDD/TTY interfaces with
the user.  From the looks of it, fskmodem talks directly to a channel.
Does it matter what type of channel it connects to?  SIP/IAX/Zap?
Secondly, how does one interface with it on the asterisk side?
Obviously there is no sendtty function in the cli and it would be the
wrong place for it.  How does it work?

Thanks
-Don

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Re: [Asterisk-Users] Delays in IVR

[Rich Adamson]

> 
> >>
> >>   ;extensions for dan and adam
> >>   ;dan - since people already know dan as extension 3, we keep
> >>that for compatibility
> >>   exten => 3,1,GoTo(Pleximenu|103|1)
> >>   exten => 103,1,GoTo(default|103|1)
> >>
> >>   ;adam
> >>   exten => 104,1,GoTo(default|104|1)
> >>
> >>
> >>
> >
> >
> > The bottom of the dialplan is your culprit here. It's waiting the
> >additional time because it's not sure whether or not you're going to
> >enter 103 or 104 as opposed to just 1, so it's waiting for the digit
> >timeout to be sure.
> >  
> >
> 
> Several people made that suggestion, but I had already tried it with 
> those extensions commented out.  Would anything be neccesary to make the 
> change take effect aside from "extensions reload"?

As someone else mentioned, have you tried playing with:
[bus-ivr-main]
exten => s,1,Wait,1 
exten => s,2,Answer 
exten => s,3,Set(TIMEOUT(digit)=5)   
exten => s,4,Set(TIMEOUT(response)=10)

and changing the digit timeout value to something different?


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RE: [Asterisk-Users] CDR_CSV stops writing, help!

[Alexander Lopez]

Tyler, 

Can you upgrade to 1.2???

Alex 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Tyler
> Sent: Tuesday, December 27, 2005 9:40 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] CDR_CSV stops writing, help!
> 
> Hi list..
> 
> Using asterisk 1.0.10 and the cdr_mysql addon to write CDR 
> records to a MySQL table.  That part works great.  The issue 
> is that I also need the Master.csv text CDR log and thusly 
> have the cdr_csv.so module loaded. 
> The problem is, after 10-15 mins of activity, it just.. stops 
> writing. 
> tail -f Master.csv no longer shows anything, restarting 
> asterisk doesn't flush any buffers, etc.  It just stops.
> 
> Issuing a reload or reloading the cdr_csv.so module gets it 
> working again for another few minutes, but I don't get why 
> it's stopping.
> 
> I've got plenty of disk space available and the Master.csv 
> file is only 500k in size (right now).
> 
> Anyone come across this before?
> 
> tf.
> 
> 
> 
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[Asterisk-Users] Cisco dtmf

[Tomislav Parcina]

I'm trying to set up call transfer and automon options. They work fine 
with ZAP lines (analog telephone) and with Grandstream Budgetone 102. I 
have problem with Cisco 7905 and 7940. I think that problem is with dtmf 
signalization. 

This is my configuration in 7940 
dtmf_inband: 1
dtmf_outofband: none
dtmf_db_level: 3

And 7905
AudioMode:0x

Is my configuration wrong or it doesn't work with Cisco phones?

Thank you for your time!


-- 

Tomislav Parcina
[EMAIL PROTECTED]

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Re: [Asterisk-Users] Extension cannot match ! receiving call mISDN ...

[Ivo Simicevic]

On 12/21/05, Joao Correia <[EMAIL PROTECTED]> wrote:
Hello,
 
Making calls works fine on a Beronet 1 port card connected to an ISDN line PTP.
I cant seam to receive any calls. Asterisk says it cannot match
extension. The funny is that I tested this configuration on a ptmp and
it worked.
Any tips ?

Maybe you are missing [misdn-in] context in extensions.conf ?

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Re: [Asterisk-Users] Unicall E1 Error in Mexico

[Martinez Felix]

no podria decirte, porqe tengo problemas con los scripts de email2fax y
Asterfax...espero resolverlos pronto y verificar el correcto envio de
faxes...On 12/21/05, Jorge Cisneros <[EMAIL PROTECTED]> wrote:
gracias Felix por el tip, ya lo hice y si funciono todo bien. tengo
otro problema no puedo enviar fax a traves de las lineas de unicall
creo que el problema esta en la cancelacion de echo. Tu has tenido este
problema

GRacias 
On 12/21/05, Martinez Felix <[EMAIL PROTECTED]
> wrote:
Es un timeout...necesitas incrementarlo...en la libreria de unicall existe un archivo qe se llama mfcr2.c...

#define BLOCKING_RELEASE_TIME   450
#define
ANSWER_GUARD_TIME  
100#define
DEFAULT_T1 
5000  <-Dale una valor mas alto...2 por ejemplo
#define
DEFAULT_T1A
150
#define
DEFAULT_T1B
6
#define
DEFAULT_T2 
5000
#define
DEFAULT_T3 
15000

vuelves a compilar y a instalar y listo...
On 12/20/05, Jorge Cisneros <

[EMAIL PROTECTED]> wrote:

Hi 
 
  I have a weired problem. when i make a call with some numer the unicall show me a error. 
 
For example when i dial 30003300 in mexico city then log show 
 
MFC/R2 UniCall/3 R2 prot. err. [2/ 
40/Group I  
/DNIS ] cause 32769 -
T1 timed outDec 21 00:22:46 WARNING[17649]: MFC/R2 UniCall/3 8 off
-> 
[1/  
1/Idle 
/Idle ]Dec 21
00:22:46 WARNING[17649]: MFC/R2 UniCall/3 1001 
-> 
[1/  
1/Idle 
/Idle ]
Dec 21 00:22:46 WARNING[17649]: Unicall/3 event Protocol failureDec 21 00:22:46 VERBOSE[17649]: -- Unicall/3 protocol error. Cause 32769 
 
But with other number work fine. The problem is only with a few number.
 
thanks
 
 

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Re: [Asterisk-Users] spandsp & fax

[Kristof Hardy]

Dov Bigio wrote:
I am using Asterisk 1.2.1 and followed instructions on 
http://www.asteriskguru.com/tutorials/spandsp.html to install faxing 
capability on my server.


what platform are you running on? (wich distro?)
Does the make of the app_txfax and app_rxfax work out well?


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[Asterisk-Users] Re: Transfer

[Tomislav Parcina]

In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
> My understanding is that canreinvite only redirects the media path. 
> Signaling and media are separate with SIP (which is what makes it so 
> nice by the way).

Yes, and dtmf can be sent with the sound. And if that is the case, then 
media path needs to go thrue *.

> Dunno. If it works at all, asterisk will have to redirect the media 
> stream to itself. Have you tried?

Yes, I have tried it and it doesn't work. Now I have found out that 
problem is with Cisco phones. I'm opening new thread - Cisco dtmf.


-- 

Tomislav Parcina
[EMAIL PROTECTED]

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RE: [Asterisk-Users] Stay away from Grandstream!

[Chris Bagnall]

> We've contact Grandstream support, but they cannot help. Now 
> we want to send devices to Grandstream for repair but they on 
> longer reply mail!

This is where a good reseller is worth their weight in gold. Unless you're
buying massive quantities of the things (in which case a failure of 2 is
pretty good by my standards) you're unlikely to be dealing with Grandstream
directly anyway. Go back to the reseller who sold you the phones and get
*them* to deal with Grandstream.

FWIW, we've got 4 sites with GXP2000s, probably a total of 50 phones now,
out of which 2 have had issues - one had a dodgy power connector that meant
the slightest movement would power cycle the phone, the other had a dodgy
off-hook button (phone wouldn't always clear down properly). All in all,
even on my relatively small sample size, that's an acceptable failure rate.

In both cases, I phoned our distributor, informed them of the fault and they
had a courier deliver replacement phones to us the following day, and
collected the offending phones at the same time, all at no cost to us or our
clients.

> GXP-2000 was very buggy on attended call transfer, and the 
> problem resolved only after upgrading using latest firmware. 
> Overall GXP is OK, but customer support is terrible. Stay 
> away from them!

What issues are you having with attended call transfer? In recent months
I've gone through a fair number of GXP2000 firmware versions and I can't say
any of them have had a problem with attended transfer.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


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Re: [Asterisk-Users] spandsp & fax

[BJ Weschke]

On 12/27/05, Dov Bigio <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I am using Asterisk 1.2.1 and followed instructions on
> http://www.asteriskguru.com/tutorials/spandsp.html to
> install faxing capability on my server.
>
> I get the following error messages...
>
> Asterisk Dynamic Loader Starting:
>   == Parsing '/etc/asterisk/modules.conf': Found
>  [app_rxfax.so]Dec 27 12:14:27 WARNING[14679]: loader.c:334 __load_resource:
> No load_module in module
> /usr/lib/asterisk/modules/app_rxfax.so
> Dec 27 12:14:27 WARNING[14679]: loader.c:341 __load_resource: No
> unload_module in module
> /usr/lib/asterisk/modules/app_rxfax.so
> Dec 27 12:14:27 WARNING[14679]: loader.c:348 __load_resource: No usecount in
> module /usr/lib/asterisk/modules/app_rxfax.so
> Dec 27 12:14:27 WARNING[14679]: loader.c:355 __load_resource: No description
> in module /usr/lib/asterisk/modules/app_rxfax.so
> Dec 27 12:14:27 WARNING[14679]: loader.c:362 __load_resource: No key routine
> in module /usr/lib/asterisk/modules/app_rxfax.so
> Dec 27 12:14:27 WARNING[14679]: loader.c:371 __load_resource: Key routine
> returned NULL in module
> /usr/lib/asterisk/modules/app_rxfax.so
> Dec 27 12:14:27 WARNING[14679]: loader.c:380 __load_resource: 6 errors
> loading module /usr/lib/asterisk/modules/app_rxfax.so,
> aborted
> Dec 27 12:14:27 WARNING[14679]: loader.c:499 load_modules: Loading module
> app_rxfax.so failed!
> [EMAIL PROTECTED] modules]#

 Which version of the app_rxfax and app_txfax applications did you
grab from http://www.soft-switch.org/?

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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Re: [Asterisk-Users] IAX media path, forcing server to stay in the middle

[Time Bandit]

> Why not just put 'notransfer=yes' in the appropriate iax.conf user/peer entry?
Oups, answered too fast. That is what happens when I try to answer a
technical question before finishing my first coffee.

Thanks for the correction
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[Asterisk-Users] Re: Transfer

[Tomislav Parcina]

In article <[EMAIL PROTECTED]
aachen.de>, [EMAIL PROTECTED] says...
> It is not only re-invite that determines what happens to your media path, 
> there are also Dial() arguments like t,T,w,W (and possibly some more) 
> that can force it go through Asterisk. The same applies to codec 
> settings, i.e. if you need Asterisk in between to transcode e.g. from 
> g729 to alaw then obviously the rtp stream has to go thru Asterisk.

Thank you for explaining this to me.

> Next to that: Try to switch both your phone and your Asterisk config to 
> dtmfmode=info (SIP INFO) and see if automon recording will work that way 
> even if you have canreinvite=yes - it could work since in this case DTMF 
> is transmitted as SIP message; I have to admit that I am not 100% sure if 
> with canreinvite=yes Asterisk will also be completely cut off from the 
> SIP signalling stream, but I think it'll still be in the loop - haven't 
> tried it myself.

I have found out that the problem is with Cisco phones (7905 and 7940). 
With ZAP (analog) phones and with Grandstream it works fine.

I'm opening new thread for this one => Cisco dtmf

> For your transfer question: You'll have to use t or T in Dial in order to 
> permit transfer, which in turn means your rtp traffic will be forced thru 
> Asterisk no matter what your canreinvite= settings looks like.
> You might want to look at if and how your SIP phone supports native 
> transfer by itself.

My phones support nativ transfer but I would like to implement this 
feature.


-- 

Tomislav Parcina
[EMAIL PROTECTED]

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Re: [Asterisk-Users] IAX media path, forcing server to stay in the middle

[Kevin P. Fleming]

Time Bandit wrote:


The trick is to use some Dial options that forces * to stay in the
path, like t,T,h,H,w or W
See http://www.voip-info.org/wiki-Asterisk+cmd+Dial for an explanation
of those options.


Or set 'notransfer=yes' for at least one of the IAX2 peers/users involved.
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[Asterisk-Users] CDR_CSV stops writing, help!

[Tyler]

Hi list..

Using asterisk 1.0.10 and the cdr_mysql addon to write CDR records to a
MySQL table.  That part works great.  The issue is that I also need the
Master.csv text CDR log and thusly have the cdr_csv.so module loaded. 
The problem is, after 10-15 mins of activity, it just.. stops writing. 
tail -f Master.csv no longer shows anything, restarting asterisk doesn't
flush any buffers, etc.  It just stops.

Issuing a reload or reloading the cdr_csv.so module gets it working
again for another few minutes, but I don't get why it's stopping.

I've got plenty of disk space available and the Master.csv file is only
500k in size (right now).

Anyone come across this before?

tf.



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Re: [Asterisk-Users] How to check Digium TE410P firmware version?

[Kevin P. Fleming]

Steve Totaro wrote:


Field-upgradeable?  Does that mean that I can do it myself?  That would
be great since some systems are in production and sending the board to
Digium takes time.


The 2nd gen firmware has field-upgradeability. The 1st gen firmware does 
not, unfortunately. There is not currently any 3rd gen firmware, but 
when there is, you'll be able to do it yourself :-)

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Re: [Asterisk-Users] IAX media path, forcing server to stay in the middle

[Andrew Kohlsmith]

On Tuesday 27 December 2005 09:18, Time Bandit wrote:
> The trick is to use some Dial options that forces * to stay in the
> path, like t,T,h,H,w or W
> See http://www.voip-info.org/wiki-Asterisk+cmd+Dial for an explanation
> of those options.

Why not just put 'notransfer=yes' in the appropriate iax.conf user/peer entry?

-A.
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[Asterisk-Users] RE: [Asterisk- Pls. explain what happens...

[Kevin Steil]

Depends if you have reinvite on or off.  On yes  off no.  At least that
is what I have read...you can verify with a network sniff on the
Asterisk server...use 

tcpdump -ln host ip.off.asterisk.server and not tcp port ssh  (telnet or
whatever protocol you are connecting to the astersisk server with)

-Original Message-
From: Mauro Zanin [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, December 27, 2005 3:27 AM
To: Asterisk User List
Subject: [Asterisk-Users] Pls. explain what happens...

Hi everybody,
can anybody explain one thing: say we have 2 SIP phones(or H323) and one
Asterisk Box on one local network. The phone1 calls phone 2 via Asterisk
and
phon3 answers: is the real conversation streaming thru the * box, or
it's
going straigth from one phone to the other?

Regards and Happy New Year.

Mauro

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