[Asterisk-Users] 1.0.10 to 1.2.1 upgrade..is it worth it?

2006-01-10 Thread stevanus

Hi,

As I've dealt with asterisk 1.0.10 successfully, I wonder what the 
benefit I will get from upgrading to 1.2.1..
[Of course I know there're lot of new interesting stuffs  in 1.2.1, but 
are they stable already?]


Does the 1.2.1 need more resources, more power hungry?

Anyone has success story with asterisk 1.2.1 please share :)

Thank you...

Regards,

Stevanus
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RE: [Asterisk-Users] Recommendations on a WiFi phone for *?

2006-01-10 Thread Ivan Meic
>> Has anyone tried out Hitachi IPC-5000 ?
>> It looks nice and it's a bit exensive, but I would still like to hear
>> how does it behave around Asterisk.
>>
>
>I used one for about 6 months. In general its well behaved with
>Asterisk. It is a little tedious to get setup, but then you only need
>to doit once. My issue with it were simple. The volume level was always
>too low. Also, in order for a call to be sustained while moving between
>access points they had to have the same SSID and be on the same
>channel. That's less than ideal. Also, it useful range from the AP was
>limited.

Can Hitachi do any "extra" features like Call Hold, Xfer, BlndXfer,
Conference maybe ?
My best choice so far (the only phone I've tested that actually works) is
ZyXEL,
but it can not do any of the features mentioned above.

Thanks,
Ivan


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Re: [Asterisk-Users] Eid Mubarak

2006-01-10 Thread Remco Barende
Is there no list moderation, could somebody please kick all these spammers 
off the list?


This just sucks, the lists volume is very high already even without the 
spam

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Re: [Asterisk-Users] Problem with Chan_zap.so

2006-01-10 Thread Dinesh Nair



On 01/10/06 04:32 Arinze Izukanne said the following:

I just upgraded to Asterisk 1.2.1 and Asterisk fails
to start with the error below.

Jan  9 21:25:38 NOTICE[1339]: cdr.c:1171 do_reload:
CDR simple logging enabled.
Jan  9 21:25:38 WARNING[1339]: loader.c:326
__load_resource:
/usr/lib/asterisk/modules/chan_zap.so: undefined
symbol: pri_restart


libpri seems to be missing from your library search path.

--
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[EMAIL PROTECTED](0 0)http://www.alphaque.com/
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| for a in past present future; do|
|   for b in clients employers associates relatives neighbours pets; do   |
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+=+
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Re: [Asterisk-Users] Incoming Zap channels not behaving as expected. Rejecting call on channel....

2006-01-10 Thread Dinesh Nair


On 01/10/06 11:06 Beau Hargis said the following:

-- Extension '2061234567' in context 'default' from '206987654' does
not exist.  Rejecting call on channel 0/16, span 4

When I add '_206XXX,1,Goto(demo,s,1)' I can get it to work.

This is going to be for an IVR application not a PBX. So, numbers are


you could have calls immediately sent to the s extension by the use of 
immediate=yes in zapata.conf. have you tried this option ?


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RE: [Asterisk-Users] VOIP Termination

2006-01-10 Thread Ross C
You need to find a VOIP provider for termination.  There are lots out there.
I use VoipJet.com.  they've been reliable and good overall.
Then you setup a new trunk in Asterisk for VoipJet.
The users would need some sort of SIP device at their premises; like a SIP
phone or an SPA-3000 or something.  There are other ways, but this is the
most generic way to set it all up.

So a user at House1 would pick up their SIP phone, dial a number, the phone
would connect with the Asterisk server, the Asterisk server passes the
number to dial to your VOIP provider (voipjet or whatever), and your voip
provider handles to VOIP-PSTN connection.  Sounds like you already have SIP
phones or something setup, so you should be OK there.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Duracom ISP
Lists
Sent: Tuesday, January 10, 2006 8:45 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] VOIP Termination

First of all I am a complete noob to VOIP and Asterisk.  We currently have
an Asterisk box setup and working with 4 FXO ports to the PSTN.  What if I
wanted to build another Asterisk box for my ISP users so they could use VOIP
instead of the PSTN.  Am I correct in saying that I need to find a VOIP
provider to peer/terminate with so I can offer service to my users such as
LD and VOIP call to the PSTN network?  Or would I need an Asterisk box in
every little town that I want to allow them to call to?



J




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RE: [Asterisk-Users] New Freelance Site for Asterisk Consultantsand Those who Need Projects Done

2006-01-10 Thread Steve Totaro
There is already one post for help in the 15 minutes it has been online
so maybe it will catch on.  

Status: Open   
Budget: $ 50-250  
Created: 1/10/2006 23:49  
Bidding Ends: 1/13/2006 23:49 (3 days left)  
Project Creator: skywifi 
Rating: (No Feedback Yet)  
Description: We had a company build a system for our client consisting
of the following:

1 - WCTE11XP
1 - TDM04B
1 - TDM02B

Basically, we have (6) incoming PSTN lines and (24) Analog Zap
extensions connected through a Zhone Channel Bank. On some incoming PSTN
calls, we receive a loud screeching noise. This is an intermittent
problem that is very irritating.

> -Original Message-
> From: Darren Wiebe [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, January 10, 2006 11:36 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] New Freelance Site for Asterisk
> Consultantsand Those who Need Projects Done
> 
> Cool!  I just tool a look at it looks like you did a great job!!
> 
> Darren Wiebe
> [EMAIL PROTECTED]
> 
> Steve Totaro wrote:
> 
> >Sorry if this is slightly off topic but it does pertain to Asterisk
> >Users as well as the biz list.  Also, sorry if it is a double post
but
> >the first one never made it to the list for some reason.
> >
> >
> >
> >Hello all,
> >
> >
> >
> >I have created a beta site for "Asterisk Gurus" or Consultants to bid
on
> >projects posted by customers needing to have work done.  It is very
> >similar to scriptlance or any of those other sites but it is
dedicated
> >to Asterisk and related issues so hopefully only really qualified
> >Asterisk consultants will bid on your projects.  If you post at one
of
> >those other sites, you wind up with 99% of the people who bid unable
to
> >complete the project and they waste your valuable time.
> >
> >
> >
> >Asterisk is a very specialized skill and with our rating system, we
can
> >quickly identify who the good "Asterisk Gurus" are and not waste time
> >with the wannabes.
> >
> >
> >
> >This also seems to be a very good replacement for the "Bounty" system
on
> >www.voip-info.org  .  I am sure we can
figure
> >out how to split costs owed to the "Asterisk Guru" between customers.
> >
> >
> >
> >It is VERY beta right now but I think it is also fully functional.
Any
> >reference to payments, deposits, $$$, etc can be ignored.  The
service
> >is free for now and will stay that way for at least the next six
months.
> >
> >
> >
> >
> >However, I am may add a PayPal donation link since this is certainly
not
> >free for me.  Of course there is no obligation to donate but I would
> >appreciate it.  Heck, if there are enough donations then the site
could
> >remain free permanently.
> >
> >
> >
> >There are some small issues since the script is wrapped in another
> >script, but I am aware of this and will find a fix shortly.  Besides
> >that, I could use any input on usability, additions, categories not
> >listed, or whatever jumps to mind.
> >
> >
> >
> >I will also be adding a section to post resumes and other permanent
job
> >postings.
> >
> >
> >
> >Please test it out and let me know what you think.
> >
> >
> >
> >http://www.asteriskhelpdesk.com
> >
> >
> >
> >Thanks,
> >
> >Steve Totaro
> >
> >
> >
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> >
> >
> 
> 
> --
> Darren Wiebe
> [EMAIL PROTECTED]
> Aleph Communications
> ASTPP - Open Source Voip Billing & Calling Cards
> www.aleph-com.net/astpp
> 
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Re: [Asterisk-Users] New Freelance Site for Asterisk Consultants and Those who Need Projects Done

2006-01-10 Thread Darren Wiebe

Cool!  I just tool a look at it looks like you did a great job!!

Darren Wiebe
[EMAIL PROTECTED]

Steve Totaro wrote:


Sorry if this is slightly off topic but it does pertain to Asterisk
Users as well as the biz list.  Also, sorry if it is a double post but
the first one never made it to the list for some reason.



Hello all,



I have created a beta site for "Asterisk Gurus" or Consultants to bid on
projects posted by customers needing to have work done.  It is very
similar to scriptlance or any of those other sites but it is dedicated
to Asterisk and related issues so hopefully only really qualified
Asterisk consultants will bid on your projects.  If you post at one of
those other sites, you wind up with 99% of the people who bid unable to
complete the project and they waste your valuable time.



Asterisk is a very specialized skill and with our rating system, we can
quickly identify who the good "Asterisk Gurus" are and not waste time
with the wannabes.



This also seems to be a very good replacement for the "Bounty" system on
www.voip-info.org  .  I am sure we can figure
out how to split costs owed to the "Asterisk Guru" between customers.



It is VERY beta right now but I think it is also fully functional.  Any
reference to payments, deposits, $$$, etc can be ignored.  The service
is free for now and will stay that way for at least the next six months.




However, I am may add a PayPal donation link since this is certainly not
free for me.  Of course there is no obligation to donate but I would
appreciate it.  Heck, if there are enough donations then the site could
remain free permanently.



There are some small issues since the script is wrapped in another
script, but I am aware of this and will find a fix shortly.  Besides
that, I could use any input on usability, additions, categories not
listed, or whatever jumps to mind.



I will also be adding a section to post resumes and other permanent job
postings.



Please test it out and let me know what you think.



http://www.asteriskhelpdesk.com



Thanks,

Steve Totaro



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--
Darren Wiebe
[EMAIL PROTECTED]
Aleph Communications
ASTPP - Open Source Voip Billing & Calling Cards
www.aleph-com.net/astpp

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[Asterisk-Users] New Freelance Site for Asterisk Consultants and Those who Need Projects Done

2006-01-10 Thread Steve Totaro
Sorry if this is slightly off topic but it does pertain to Asterisk
Users as well as the biz list.  Also, sorry if it is a double post but
the first one never made it to the list for some reason.

 

Hello all,

 

I have created a beta site for "Asterisk Gurus" or Consultants to bid on
projects posted by customers needing to have work done.  It is very
similar to scriptlance or any of those other sites but it is dedicated
to Asterisk and related issues so hopefully only really qualified
Asterisk consultants will bid on your projects.  If you post at one of
those other sites, you wind up with 99% of the people who bid unable to
complete the project and they waste your valuable time.

 

Asterisk is a very specialized skill and with our rating system, we can
quickly identify who the good "Asterisk Gurus" are and not waste time
with the wannabes.

 

This also seems to be a very good replacement for the "Bounty" system on
www.voip-info.org  .  I am sure we can figure
out how to split costs owed to the "Asterisk Guru" between customers.

 

It is VERY beta right now but I think it is also fully functional.  Any
reference to payments, deposits, $$$, etc can be ignored.  The service
is free for now and will stay that way for at least the next six months.


 

However, I am may add a PayPal donation link since this is certainly not
free for me.  Of course there is no obligation to donate but I would
appreciate it.  Heck, if there are enough donations then the site could
remain free permanently.

 

There are some small issues since the script is wrapped in another
script, but I am aware of this and will find a fix shortly.  Besides
that, I could use any input on usability, additions, categories not
listed, or whatever jumps to mind.

 

I will also be adding a section to post resumes and other permanent job
postings.

 

Please test it out and let me know what you think.

 

http://www.asteriskhelpdesk.com

 

Thanks,

Steve Totaro

 

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[Asterisk-Users] SOLVED: Hung Zap channels connected to old key system

2006-01-10 Thread Philip Edelbrock


We've got a Toshiba DK system w/ analog ports that went to a  
voicemail server.  I swapped in an Asterisk box with a Digium 4-port  
fxo card.  It /almost/ worked perfectly.


The problem is that Zap channels never hang up.  They have to time out.

I set up MeetMe, but all Zap channels hung forever.  Very annoying.   
Same thing for FXO-to-FXO bridges.


I figured out today why and fixed it.  Some proprietary voicemail  
systems (and probably tie-lines, too) like to use DTMF tones instead  
of standard ground/loop/kewl whatever signaling.  Our key system was  
programmed to use such DTMF tones instead of the usual analog  
signaling on those ports. (I think it was program 31 on our Toshiba  
DK40i)  Asterisk of course ignored those, but the other systems used  
those for line signaling (including our previous 3rd party system).


Amusingly, I know now why for years we kept hearing loud DTMF tones  
when our branch office picked up their phones.  Their system, too,  
was configured to have those analog lines be connected to a voicemail  
system and not to a FXO port on a T1 CSU.



Phil
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Re: [Asterisk-Users] Eid Mubarak

2006-01-10 Thread Mark Phillips
It has to be said that Eid is a funny and possibly suspect celebration 
though.


As I understand it (from one of my Muslim underlings) 3 Mad Mulahs have 
to look for a particular phase of the moon. When they see this phase 
they declare the start of Eid. They apparently get 3 nights in which to 
look for this moon phase. I guess my question is "what happens if its 
cloudy on all 3 nights?"


Another thing I thought about is this; If we could get the Faithfull 
whom are attending the Haaj this week to suddenly apply their brakes do 
you think they could stop the world from turning? Better yet if they all 
jumped into the air at once would the resultant landing knock us off off 
our regular orbit?


Talk about "death to Ifidels"! They could do it in one fell swoop! I 
wonder if Al Quaeda has spent any research money on this?


Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com


Rusty Dekema wrote:
On 1/10/06, *Carlos Alperin* <[EMAIL PROTECTED] 
> wrote:


And, as I said before, I'm not a religious man, but I don't like other
people trying to be funny with somebody else traditions or believes.


Personally, I like being funny about traditions and beliefs a whole lot 
better than I like being overly serious (or worse yet petty) about them!


I have never heard of Eid Mubarak, so I thought it was kind of 
interesting to learn a little bit about it.


But if I were to bring up a religious subject in a technology forum (or 
list), I would consider myself to have gotten off rather lightly if the 
"worst" response I got was simply a funny little play on words such as 
the "got my goat" comment.


Now, if someone came on here and started a "I believe in Religion X and 
you believe in Religion Y, so YOU ARE DAMNED TO HELL FOR ALL ETERNITY" 
spiel or on the other hand started a "DON'T YOU (so-and-so) IDIOTS KNOW 
THERE'S NO GOD" spiel, then sure; that would be annoying and offensive. 
But nobody did that.


I don't understand why people get so worked up about this kind of thing. 
To me, the idea of you being offended by the "got my goat" joke makes 
about the same amount of sense as the idea that someone would have been 
offended by Rehan's  original post that started this thread. Although 
one person might not like to hear jokes about religion, other people 
might not like to hear about religion at all. I think that either way 
it's not worth getting particularly upset over.


-Rusty





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[Asterisk-Users] VOIP Termination

2006-01-10 Thread Duracom ISP Lists
First of all I am a complete noob to VOIP and Asterisk.  We currently have
an Asterisk box setup and working with 4 FXO ports to the PSTN.  What if I
wanted to build another Asterisk box for my ISP users so they could use VOIP
instead of the PSTN.  Am I correct in saying that I need to find a VOIP
provider to peer/terminate with so I can offer service to my users such as
LD and VOIP call to the PSTN network?  Or would I need an Asterisk box in
every little town that I want to allow them to call to?



J




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Re: [Asterisk-Users] Eid Mubarak

2006-01-10 Thread Rusty Dekema
On 1/10/06, Carlos Alperin <[EMAIL PROTECTED]> wrote:
And, as I said before, I'm not a religious man, but I don't like otherpeople trying to be funny with somebody else traditions or believes.Personally, I like being funny about traditions and beliefs a whole lot better than I like being overly serious (or worse yet petty) about them! 
I have never heard of Eid Mubarak, so I thought it was kind of interesting to learn a little bit about it. But if I were to bring up a religious subject in a technology forum (or list), I would consider myself to have gotten off rather lightly if the "worst" response I got was simply a funny little play on words such as the "got my goat" comment. 
Now, if someone came on here and started a "I believe in Religion X and you believe in Religion Y, so YOU ARE DAMNED TO HELL FOR ALL ETERNITY" spiel or on the other hand started a "DON'T YOU (so-and-so) IDIOTS KNOW THERE'S NO GOD" spiel, then sure; that would be annoying and offensive. But nobody did that. 
I don't understand why people get so worked up about this kind of thing. To me, the idea of you being offended by the "got my goat" joke makes about the same amount of sense as the idea that someone would have been offended by Rehan's  original post that started this thread. Although one person might not like to hear jokes about religion, other people might not like to hear about religion at all. I think that either way it's not worth getting particularly upset over. 
-Rusty
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RE: [Asterisk-Users] Re: Re: Remotely reboot SIP Phones ?

2006-01-10 Thread kevin ling
Thanks a lot. It's work :-) 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel
Sent: Tuesday, January 10, 2006 10:18 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] Re: Re: Remotely reboot SIP Phones ?

Figured it out :)

Basically, you have to have a file called syncinfo.xml in the tftp root
directory, with the following contents:





Also, in SIPDefault.cnf or the phone's configuration file, stick:

sync: "0"

somewhere so the phone's sync value doesn't match the value in syncinfo.xml.

If you make a change of sorts, just run "sip notify reboot-cisco "
at any time in asterisk and it'll send the notify to the phone.

If the phone is in use, it waits until it's idle, once it is, it waits 20
seconds and then checks the syncinfo.xml file, and if the values of sync are
different, it reboots :)

Aaron

Tomislav Parcina wrote:
> In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
>> Yeah, that should theoretically work, but I've got about 60 cisco 
>> phones that don't respond to the check-sync.
> 
> If you ever make it work, please anounce it on the group.
> 
> 
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RE: [Asterisk-Users] Pri Gateway Hardware

2006-01-10 Thread Jason Kim
Does TDMoE supports kernel 2.6?
Where should I do echo cancellation?

--- Carlos Alperin <[EMAIL PROTECTED]> wrote:

> Low level requeriment, just you transfer everything
> using level 2. So you
> don't need to the overhead to have Asterisk running
> to route that traffic.
> 
> Carlos Alperin
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On
> Behalf Of Jean-Michel
> Hiver
> Sent: Tuesday, January 10, 2006 1:18 PM
> To: Asterisk Users Mailing List - Non-Commercial
> Discussion
> Subject: Re: [Asterisk-Users] Pri Gateway Hardware
> 
> Alexander Lopez a ?rit :
> 
> >TDMoE is stable and stale, There is no more
> development planed or needed as
> it only opens up a pipe between two ethernet points
> using Layer 2.
> >  
> >
> OK... What would be in the advantage in running
> TDMoE rather than using 
> IAX or SIP?
> 
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Re: [Asterisk-Users] Second edition of my * book has been released

2006-01-10 Thread Gabriel Sartor
 
How much is this book ?? 
2006/1/10, Randy Williams <[EMAIL PROTECTED]>:
Greetings All,I have found that Paul's book is just right for rounding out the edgeswhen getting started.  I managed to temporarily migrate our T-1 Asterisk
system to a Analog asterisk system on information in Paul's book alone.Nicely done and a neat bit of help in a pinch.Just my $0.02 USD you understand. :)RandyWPaul Mahler wrote:>Hi Greg,
>>My book is a good place for a beginner to get started. I also find it to be>useful as a reference for Asterisk. It's not an advanced book, there are>advanced features it doesn't cover, for example AGI or the management
>interface.>>It should be very helpful for your customers. It should be helpful for a>beginning to intermediate administrator. I still frequently refer to it>myself when I'm having a senior moment. :)
>>There isn't anything in the book that would make it less useful for the CVS>or stable branches.>>The O'Reilly book is excellent. I think my book complements the O'Reilly>book. If I were just starting I would buy both. I think my book may be a bit
>more useful as a reference. I think I cover a bit more beginner's territory.>>>Hope This Helps,>>Paul>-Original Message->>From: 
[EMAIL PROTECTED] [mailto:asterisk-users->>[EMAIL PROTECTED]
] On Behalf Of [EMAIL PROTECTED]>>Sent: Monday, January 09, 2006 9:10 PM>>To: asterisk-users@lists.digium.com
>>Subject: RE: [Asterisk-Users] Second edition of my * book has been>>releasedHow does it compare with the O'Rielly book?Does it include information on CVS, or primarily on stable?
Can it be provided to customers, or is it more sysadmin oriented?Regards,>>Greg-Original Message->>From: 
[EMAIL PROTECTED]>>[mailto:[EMAIL PROTECTED]] On Behalf Of Paul>>Mahler>>Sent: Thursday, January 05, 2006 9:45 AM
>>To: 'Asterisk Users Mailing List - Non-Commercial Discussion'>>Subject: [Asterisk-Users] Second edition of my * book has been releasedThe second edition of my Asterisk book "VoIP Telephony with Asterisk" is
>>now in print. It's reorganized and expanded.TKSPaul Mahler>>Paul Mahler>>[EMAIL PROTECTED]
www.signate.com>>___>>--Bandwidth and Colocation provided by 
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Re: [Asterisk-Users] MTU and Voice Delay (latency??)

2006-01-10 Thread Rich Adamson



Absolutely not. The MTU is the Maximum Transmission Unit, and sip
packets are about 214 bytes in size (including all pkt headers). Way
smaller then the MTU.



If the only thing on my network are these Cisco Phones, would lowering the
MTU encourage more efficient transfer of data as per here:


No. The reason is that "if" the phones are the only thing on this, the 
size of the sip packets will never be greater then 214 bytes.  The mtu 
value would need to be set to something smaller then 214 bytes in order 
for the parameter to have any impact whatsoever. If you selected 150 
bytes (for example only), the sip/rtp packets would be fragmented into 
smaller pieces forcing the destination device to reassemble those 
fragments. (I'm not sure, but I'd have to guess the Cisco's don't even 
support udp fragmentation.) Fragmentation is generally considered a 
high-cost overhead and should be avoided when possible.


Given your table below, there "are" other devices on your network and 6% 
 of those are sending packets of in the 512 to 1023 byte range. If you 
set an mtu of 512, then _those_ devices would be forced to fragment 
larger packets, giving the sip/rtp packets a greater chance of getting 
onto the external network. If your sdsl circuit is running 256kb/s, an 
mtu value of 512 would generate "at best" a 16 millisecond benefit 
roughly 6% of the time. At all other times it would generate no 
improvement whatsoever. If the sdsl circuit is operating at a speed 
greater then 256k, the best case improvement would be smaller (eg, 8 ms 
for a 512k circuit).



http://www.voiptroubleshooter.com/problems/mtu.html
http://www.opalsoft.net/qos/VoIP.htm


Here is the breakdown of packets on the port that has the SDSL modem:

SDSL Router
Amount  % of Whole
64 BytePkts 0   0%
65-127 BytePkts 20257   1%
128-255 BytePkts3623518 91%
256-511 BytePkts237087  6%
512-1023 BytePkts   112900  3%
1024-1522 BytePkts  48  0%
Total   3993810 100%

The larger packets I'm sure are the bootloader stuff and config file
downloads, etc.


That's very possible.


I do feel like I am reaching for straws here!


You're reaching for _air_ by even thinking about mtu as an issue. In 20+ 
years of professional network management and performance assessments, 
the only time mtu is ever brought up, its generally by some first-year 
technical type that doesn't have a clue (that's not directed to you 
either). (FWIW, mtu adjustments use to be fairly popular for those 
attempting to balance response time vs throughput on 28kb dialup circuits.)


If you don't believe any of the above, go ahead and change the mtu and 
see for yourself. Bet the phones don't even work at all.



Just as an update, the users used to be on two 2mb down/512 up ADSL lines
(PPPoE) (4 users on each) and they never reported a problem. Now that they
are on one SDSL (PPPoA) line (2mb) is when they report the issues.


There is something else going on besides adsl vs sdsl, and mtu is not 
even close to being the root cause.


Have you tried the previous suggestion relative to two simultaneous ftp 
sessions?


What city/state are you located in?

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Re: [Asterisk-Users] Re: web sip client

2006-01-10 Thread Rehan AllahWala
I think babar wrote a sip version also


> Rehan AllahWala wrote:
> 
> >The guy who wrote it is
> >
> >Babar Shafiq  >
> >Price i belive is 500$ for it
> >
> >Babar do comment !
> >
> >Rehan
> >
> >  
> >
> It looks very good, however im looking for a sip solution instead of
> iax, another option? thanks
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Super Technologies Inc., Pensacola, Florida
http://www.SuperTec.com - Technologies from tomorrow, Today!

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Re: [Asterisk-Users] spandsp, rxfax, TDM400/zaptel, missed frames, any help?

2006-01-10 Thread Lee Howard

Ben Fried wrote:


On 1/9/06, Adam Goryachev <[EMAIL PROTECTED]> wrote:
 


I'd strongly suggest that people having problems, and who are interested
in marketing this as a solution, do some testing with different
motherboards/systems/etc and find a combination that works. Once you
have that, simply replicate as needed.


What motherboards, CPUs, etc, do you have in the systems that are
working properly?



motherboard: ASUS A7N8X-X
processor: AMD Athlon XP 2000+
fxo card: AMI-IA92/IE92 winmodem (X100P)
kernel: Linux 2.6.12.2
configuration: ACPI using IO-APIC, fxo card uses IRQ 17 alone

Lee.
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[Asterisk-Users] register to a peer register => from database

2006-01-10 Thread Rehan AllahWala
Dear All,

How to register a peer register > command in [general] area of sip.conf via 
database.


IE i have in my sip.conf 

register => 22100140:[EMAIL PROTECTED]:5060/121222100140

Where do i take it in the Database ? 


Rehan

Super Technologies Inc., Pensacola, Florida
http://www.SuperTec.com - Technologies from tomorrow, Today!

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Re: [Asterisk-Users] spandsp, rxfax, TDM400/zaptel, missed frames, any help?

2006-01-10 Thread Ben Fried
On 1/9/06, Adam Goryachev <[EMAIL PROTECTED]> wrote:
> On Mon, 2006-01-09 at 11:40 -0600, Rich Adamson wrote:
> > > It would be very interesting to know the real numbers that have it 
> > > working.
> > > The archives (and about two/three years of attempting to help others with
> > > the exact same problem) suggests no better then maybe one in ten or twenty
> > > will ever get spandsp to work with the digium x100p or TDM card.
> > >
> > > Maybe the trick is for you to identify 'exactly' which winmodem card
> > > does work; others would be very happy to give it a try without a doubt!
> >
> > > I bought a really cheap clone off ebay and it worked first time every
> > > time (so far).  Even on longer many page faxes.
> >
> > > Getting one with an md3200 chipset (which is not what I have) and will
> > > try it on that and see and report back)..  this isnt a critical system
> > > especially for faxing, so it would be interesting for me to see if this
> > > problem you are talking about has any problems with either card.  I can
> > > try to get the exact chip I have when I open it up to install the new
> > > card when it gets here)
> >
> > It would _very_ interesting to see the data, so please do post it to the
> > list.
> >
> > There are a fair number of people interested in selling small pbx's with
> > fax/modem/pos support that actually works reliably.
> >
> > Have you tried the TDM card yet?
>
> I think at the end of the day all of this has more to do with the
> motherboard and the software rather than the TDM card. I have it working
> fine with a TE410p + TDM31B cards... ie, EFTPOS connects to FXS on TDM
> which connects to the TE410p card onto the PSTN... While this may not be
> a 'small office' scenario, it shows that at least the TDM card is
> working perfectly in that system.
>
> I also use spandsp with a TE410p card in a different system, which seems
> to work perfectly as well.
>
> I'd strongly suggest that people having problems, and who are interested
> in marketing this as a solution, do some testing with different
> motherboards/systems/etc and find a combination that works. Once you
> have that, simply replicate as needed.
>
> Regards,
> Adam

What motherboards, CPUs, etc, do you have in the systems that are
working properly?

Ben
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Re: [Asterisk-Users] Incoming PSTN Calls - Can't interrupt Main Menu

2006-01-10 Thread KokMeng Loh

Hi Aisling,

You're missing the 'WaitExten' directive after playing the background 
sound file. Your lines should be like this:


[incomingpstn]
exten => s,1,Wait(1)
exten => s,n,Background(MainMenu)
exten => s,n,WaitExten(10)
exten => 1,1,Goto(internalExt,s,1)
exten => 2,1,Goto(mainconfmenu,s,1)


-kokmeng.

Aisling wrote:


Hi,

Thanks to both Iqbal and Kokmeng for the replies. 


Kokmeng I tried what you suggested however no luck...

What I have done which is currently working(kind of) is that in my
sip.conf in the [general] section I have set context=incomingpstn. My
register line looks like:

register => username:[EMAIL PROTECTED]/

In my extensions.conf I then have

[incomingpstn]
exten => s,1,Wait(1)
exten => s,n,Background(MainMenu)
exten => 1,1,Goto(internalExt,s,1)
exten => 2,1,Goto(mainconfmenu,s,1)

[internalExt]
exten => s,n,Background(InternalExtension)

[mainconfmenu]
exten => s,n,Background(MainConfMenu)

I can hear the MainMenu sound file being played. What's strange is that
when I press '1' to interrupt, which in my logic should invoke the
internalExt context, nothing happens. The MainMenu sound file continues
to play and finally I get the error:

Warning: pbx.c:2405 __ast_pbx_run: Timeout, but no rule 't' in context
'incomingpstn'

I used the 'Goto' as Iqbal suggested instead of includes...

Has anyone ever experienced this kind of behaviour before?

Many thanks,
Aisling.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of KokMeng
Loh
Sent: 09 January 2006 08:53
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Incoming PSTN Calls - Stumped

Hi,

The hostname that you used in your register directive ('provider.ie') 
must have a corresponding section in sip.conf. In your example, you used


'[provider-in]'. If that is what you actually used, then this might 
explain why your incoming goes to the default context because it 
couldn't find its own "section". Try renaming '[provider-in]' to 
'[provider.ie]'.


-kokmeng.

Aisling O'Driscoll wrote:

 


Hi,

Yes InternalExtension is the context and 2093 the extension.

Just to explain something odd that's happening (and I'm very stumped
with this)..I think my contexts are definately the reason that I
can't interrupt the menu for incoming pstn calls to choose a submenu:

My users register with my sip proxy (SER). Therefore when I create an
entry for them in sip.conf I set only one context. Also to allow for
incoming calls from my provider it seems I must direct the calls
firstly to a 'dummy' extension.

sip.conf

register => username:[EMAIL PROTECTED]/2093

[provider-in]
type=peer
host=sip.provider.ie
context=onecontext

[2092]
type=peer
other stuff
context=onecontext

So the dummy extension here is '2093' and 2092 is a phone who
registers with SER and when SER redirects to Asterisk uses the
'onecontext' context.

Now in my extensions.conf 'onecontext' includes other contexts. This
is how I get access to conference calls, creating IVR menus etc. Also
the main purpose of 'onecontext' is to allow outgoing access to the
PSTN.

[onecontext]
include => createmenu//creating an IVR menu
include => createconf//creating a conf call
etc
include => default   //used for voicemail

[createmenu]
;does something

[createconf]
;does something

;outgoing calls - main purpose of onecontext
exten => _X.,1,Dial(SIP/[EMAIL PROTECTED])
exten => _X.,2,Hangup

[default]

;mailbox for 2092 and other users


Now this is where the problems start! For incoming calls I tried to
do "include => incomingpstn" in 'onecontext' which I thought would
call a new context called 'incomingpstn' which would have an entry
for the dummy user. i.e.

[incomingpstn]

exten => 2093,1,Wait(1)
exten => 2093,n,Background(MainMenu)
exten => 1,1,Goto(InternalExtension,2093,1)//directs to another
context called Internal Extension

I also changed the [provider-in] for context=incomingpstn in my
sip.conf. However this didn't work and I kept getting directed to the
voicemail of my pstn provider. The ONLY way I could get the incoming
calls working was to add the contents of the 'incomingpstn' context
to the default context i.e.

[default]

exten => 2093,1,Wait(1)
exten => 2093,n,Background(MainMenu)
exten => 1,1,Goto(InternalExtension,2093,1)//directs to another
context called Internal Extension

With this I can hear the MainMenu when I dial my DDI but I can't seem
to interrupt to divert to another submenu. In the testing that I have
done the user that is making the call is 2092 registered with SER. If
I change the context of 2092 directly in sip.conf to incomingpstn,
then I can hear the menu and interrupt to go to the submenu. But
obviously then I don't have access to the other features in Asterisk.
The point is that I'm stumped as to why it only works in the default
context and if this is the case how do I get it to call the submenu.

This is what comes up on my asterisk co

Re: [Asterisk-Users] Pri Gateway Hardware

2006-01-10 Thread Mike Fedyk

Jean-Michel Hiver wrote:


Alexander Lopez a écrit :

TDMoE is stable and stale, There is no more development planed or 
needed as it only opens up a pipe between two ethernet points using 
Layer 2.
 

OK... What would be in the advantage in running TDMoE rather than 
using IAX or SIP?


TDMoE should allow for simpler firmware as it allows Asterisk to handle 
all of the details and just handles transferring the TDM data and 
failover in case of Asterisk server failure.


It's about the same price as a TE411P and doesn't take a slot in your 
server(s).

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Re: [Asterisk-Users] Cisco phones 7940

2006-01-10 Thread Michael Loftis



--On January 9, 2006 2:47:34 PM -0600 Aaron Daniel <[EMAIL PROTECTED]> wrote:


I know this isn't a specifically asterisk question, but does anyone know
how to make the phone NOT use it's old config?  I'm trying to get rid of
the second line registration crap and it's not working.


into the phones .cnf file

_name, _authname, and _password are the minimum.

line2_name: UNPROVISIONED
line2_authname: UNPROVISIONED
line2_shortname: UNPROVISIONED
line2_password: UNPROVISIONED
line2_displayname: "UNPROVISIONED"

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RE: [Asterisk-Users] Eid Mubarak

2006-01-10 Thread Carlos Alperin
No,

I never said that. I'm only not joking with another people believes. 

You know, your freedom can be infinite, but his too. So, I believe that in
live the respect is our own limit.

And, as I said before, I'm not a religious man, but I don't like other
people trying to be funny with somebody else traditions or believes.

So, what a heck inquisition or happy asterisk has to do with technical
issues?

Damn

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel
Hiver
Sent: Tuesday, January 10, 2006 6:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Eid Mubarak

Carlos Alperin a écrit :

>Let's put this in perspective.
>
>I'm not a believer, but I don't think that other people religion has to be
>taken for joke.
>  
>
Is this the inquisition or what? If Rehan can say "happy eid mubarak", 
why can't I say "happy asterisk"? This is an asterisk mailing list after 
all, asterisk-damn-it!

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Re: [Asterisk-Users] Still an open Seat in London for Next Weeks Signate intro to Asterisk Course

2006-01-10 Thread Mike Fedyk

It might be easier to dig instead.

[EMAIL PROTECTED] wrote:


I would love to be there, but it's just too far to drive.

regards,

PaulH

- Original Message - 
From: <[EMAIL PROTECTED]>

To: ; 
Sent: Wednesday, January 11, 2006 7:24 AM
Subject: [Asterisk-Users] Still an open Seat in London for Next Weeks
Signate intro to Asterisk Course


 


We still have a seat open in our Asterisk training course next week in
London. You can find more information at our Web site, www.signate.com

I'm going to be teaching the class.

Paul

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Re: [Asterisk-Users] Re: web sip client

2006-01-10 Thread Miguel

Rehan AllahWala wrote:


The guy who wrote it is

Babar Shafiq  

It looks very good, however im looking for a sip solution instead of 
iax, another option?
thanks 
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Re: [Asterisk-Users] Recommendations on a WiFi phone for *?

2006-01-10 Thread pdhales
My opinion is that whoever gets a WIFI SIP phone right (and I mean really
right) is going to be a rich person.

PaulH

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

Sent: Wednesday, January 11, 2006 9:51 AM
Subject: Re: [Asterisk-Users] Recommendations on a WiFi phone for *?


>
> Lots of great comments.  Got me doing a lot of researching and thinking.
>
> What I love is the ability to mix and match IAX and SIP devices on the
> same system.  A breath of fresh air compared to our existing Toshiba key
> system.
>
> That said, I think we'll be doing a combination of things.  Regular SIP
> hardphones, atas, and sip wifi phones.  I'm still a bit stuck on the
> wifi phones because we can have folks use them in the office, the coffee
> shop, and at home (where ever wifi is, basicly).  Cost is pretty
> similar, really, compared to an ata and cordless setup (approx $200
> either way).
>
> I might wait for the F3000.  It seems to support better security,
> networking (g), and has a nicer profile.
>
>
> Phil
>

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Re: [Asterisk-Users] Eid Mubarak

2006-01-10 Thread Jean-Michel Hiver

Carlos Alperin a écrit :


Let's put this in perspective.

I'm not a believer, but I don't think that other people religion has to be
taken for joke.
 

Is this the inquisition or what? If Rehan can say "happy eid mubarak", 
why can't I say "happy asterisk"? This is an asterisk mailing list after 
all, asterisk-damn-it!


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RE: [Asterisk-Users] Recommendations on a WiFi phone for *?

2006-01-10 Thread The VoIP Connection
We sell this phone and I like it a lot, but I think Paul is right about
wireless in a typical office environment.  If, however, you want a phone
that you can use in wireless hotspots OR if your office has great 802.11
infrastructure OR if your boss likes to show off his gadgets, then the Zyxel
P2000W is a great choice. -Mike

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

> -Original Message-
> From: Paul Mahler [mailto:[EMAIL PROTECTED] 
> Sent: Tuesday, January 10, 2006 3:42 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Recommendations on a WiFi phone for *?
> 
> I would be MUCH more tempted to use an IAXy or SIP adaptor 
> and a cordless phone. It will be less expensive and it will 
> likely work better. 
> 
> Paul
> 
> > -Original Message-
> > From: [EMAIL PROTECTED] 
> [mailto:asterisk-users- 
> > [EMAIL PROTECTED] On Behalf Of Joash Herbrink
> > Sent: Monday, January 09, 2006 11:53 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: RE: [Asterisk-Users] Recommendations on a WiFi phone for *?
> > 
> > The zyxel p2000W
> > Works fine, good batt. Live.
> > Decent sound quality.
> > 
> > All in all a good product for about 150 euro's
> > 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf 
> Of Philip 
> > Edelbrock
> > Sent: Tuesday, January 10, 2006 2:45 AM
> > To: asterisk-users@lists.digium.com
> > Subject: [Asterisk-Users] Recommendations on a WiFi phone for *?
> > 
> > 
> > We're getting our feet more and more wet with VOIP at work. 
>  We want 
> > to experiment with a good wireless (as in WiFi) phone.  
> What would be 
> > a good phone to impress my boss with?
> > 
> > I'm personally drooling over the UTStarcom F3000, but compatibility 
> > and shipping ETA info is a bit sketchy.
> > 
> > 
> > Phil
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> > 
> > --
> > No virus found in this incoming message.
> > Checked by AVG Free Edition.
> > Version: 7.1.371 / Virus Database: 267.14.15/223 - Release Date: 
> > 1/6/2006
> 
> 
> 
> 

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Re: [Asterisk-Users] Recommendations on a WiFi phone for *?

2006-01-10 Thread Philip Edelbrock


Lots of great comments.  Got me doing a lot of researching and thinking.

What I love is the ability to mix and match IAX and SIP devices on the 
same system.  A breath of fresh air compared to our existing Toshiba key 
system.


That said, I think we'll be doing a combination of things.  Regular SIP 
hardphones, atas, and sip wifi phones.  I'm still a bit stuck on the 
wifi phones because we can have folks use them in the office, the coffee 
shop, and at home (where ever wifi is, basicly).  Cost is pretty 
similar, really, compared to an ata and cordless setup (approx $200 
either way).


I might wait for the F3000.  It seems to support better security, 
networking (g), and has a nicer profile.



Phil


Paul Mahler wrote:

I would be MUCH more tempted to use an IAXy or SIP adaptor and a cordless
phone. It will be less expensive and it will likely work better. 


Paul



-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Joash Herbrink
Sent: Monday, January 09, 2006 11:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Recommendations on a WiFi phone for *?

The zyxel p2000W
Works fine, good batt. Live.
Decent sound quality.

All in all a good product for about 150 euro's

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philip
Edelbrock
Sent: Tuesday, January 10, 2006 2:45 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Recommendations on a WiFi phone for *?


We're getting our feet more and more wet with VOIP at work.  We want to
experiment with a good wireless (as in WiFi) phone.  What would be a
good phone to impress my boss with?

I'm personally drooling over the UTStarcom F3000, but compatibility and
shipping ETA info is a bit sketchy.


Phil
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No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.371 / Virus Database: 267.14.15/223 - Release Date: 1/6/2006




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Re: [Asterisk-Users] Recommendations on a WiFi phone for *?

2006-01-10 Thread Mojo with Horan & Company, LLC
And possibly MUCH farther from the base antenna.  Seems like you'd 
really have to soup up your access point's signal strength and antenna 
to match the range some of the new cordless phones have.  I can go 
further on my way cheaper 2.4ghz panasonic cordless (which, in fact, 
uses an IAXy, as Paul suggested, and comes with two handsets) than I can 
with my Zyxel 2000W, and I have a 4-foot omni antenna plugged into a 
Metrix box with a fairly strong (200mW?) minipci card.




Paul Mahler wrote:

I would be MUCH more tempted to use an IAXy or SIP adaptor and a cordless
phone. It will be less expensive and it will likely work better. 


Paul



-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Joash Herbrink
Sent: Monday, January 09, 2006 11:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Recommendations on a WiFi phone for *?

The zyxel p2000W
Works fine, good batt. Live.
Decent sound quality.

All in all a good product for about 150 euro's

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philip
Edelbrock
Sent: Tuesday, January 10, 2006 2:45 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Recommendations on a WiFi phone for *?


We're getting our feet more and more wet with VOIP at work.  We want to
experiment with a good wireless (as in WiFi) phone.  What would be a
good phone to impress my boss with?

I'm personally drooling over the UTStarcom F3000, but compatibility and
shipping ETA info is a bit sketchy.


Phil
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--
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.371 / Virus Database: 267.14.15/223 - Release Date: 1/6/2006




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--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
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[Asterisk-Users] TDM04B odd problem

2006-01-10 Thread Jerry Geis

I have a TDM04B card. had been working for a while.
using head from aug 2 2005. I just updated to 1.2.1 and everything is 
fine except 1 thing.

Line 1&2 is company A line 3&4 company B. Call comes in on line 1 transfers
to SIP/205 no problem. Call comes in on line 3 attempts to also transfer 
to SIP/205

and does not work. 205 hears the ring. They pickup and dead air.
The calling in person just continiues to get ring like the line was 
never picked up.


the SIP/205 is a uniden UIP-200. The 2 companies are the same just 
answer the phone in a differnt

way. all extensions are shared etc... sip.conf has nat=never and qualify=no.

Any idea why it might do that?

I briefly switched back to head from aug 2005 and it is also doing the 
same thing.


I have not physically moved any lines yet? I am not at that location.

jerry
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Re: [Asterisk-Users] VizuFon CIP-4500 with Asterisk through SIP

2006-01-10 Thread Ian White


Make sure you have a recent copy of the firmware. There was a bug  
preventing registrations from succeeding until Nov 08 2005 and newer  
firmwares.


--
Ian White
South Island Community Access Network (SICAN)
email: [EMAIL PROTECTED]
http://sican.tc.ca/

On Dec 13, 2005, at 17:24, [EMAIL PROTECTED] wrote:



Anybody got already to make Vizufon CIP-4500 working with Asterisk  
through SIP?


I got to register by Asterisk send a "Notify" back and receive a  
"Bad Request"


Isamar


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RE: [Asterisk-Users] MTU and Voice Delay (latency??)

2006-01-10 Thread Geoff Manning
Rich Adamson wrote:
> Absolutely not. The MTU is the Maximum Transmission Unit, and sip
> packets are about 214 bytes in size (including all pkt headers). Way
> smaller then the MTU.
> 

If the only thing on my network are these Cisco Phones, would lowering the
MTU encourage more efficient transfer of data as per here:

http://www.voiptroubleshooter.com/problems/mtu.html
http://www.opalsoft.net/qos/VoIP.htm


Here is the breakdown of packets on the port that has the SDSL modem:

SDSL Router
Amount  % of Whole
64 BytePkts 0   0%
65-127 BytePkts 20257   1%
128-255 BytePkts3623518 91%
256-511 BytePkts237087  6%
512-1023 BytePkts   112900  3%
1024-1522 BytePkts  48  0%
Total   3993810 100%

The larger packets I'm sure are the bootloader stuff and config file
downloads, etc.

I do feel like I am reaching for straws here!

Just as an update, the users used to be on two 2mb down/512 up ADSL lines
(PPPoE) (4 users on each) and they never reported a problem. Now that they
are on one SDSL (PPPoA) line (2mb) is when they report the issues.
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Re: [Asterisk-Users] TE405p -- loopback for the phone company?

2006-01-10 Thread Kevin Bockman

Eric Lyons wrote:

Feeling once again like an idiot, I come to the list for help...

While my carrier (MCI vi PRI, fwiw) was trying to diagnose a problem, 
they asked me to put my interface in loopback -- and I couldn't figure 
out what they meant or how to do it.  I've plugged a loopback 
*connector* into my 405p and tested it internally, but they wanted 
something that, apparently, looped back _their_ signal.


What the?

I got zttool running and selected "loop" on the interface, but it didn't 
seem to do what they wanted (nor could I tell that it did anything at 
all).  Many googles for zaptel and loop didn't turn up anything useful.


Yeah, I tried to do that before too but it didn't do anything.  It must 
not support all switch types or something.  The best you can do probably 
is to loop it at another place, but it gets rid of testing the equipment :(


Remote loopup isn't supported either.


Kevin
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RE: [Asterisk-Users] Eid Mubarak

2006-01-10 Thread Carlos Alperin
Let's put this in perspective.

I'm not a believer, but I don't think that other people religion has to be
taken for joke.

Rehan, Happy Eid Mubarak (however I believe that this is not a right place
for publish this)

Jean-Michael, Happy Asterisk with a better respect for the rest of the
users.

Carlos Alperin

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel
Hiver
Sent: Tuesday, January 10, 2006 3:37 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Cc: asterisk-biz@lists.digium.com
Subject: Re: [Asterisk-Users] Eid Mubarak

Rehan AllahWala a écrit :

>Dear All,
>
>For those who celebrate Eid.
>
>I would like to wish you a very Happy Eid Mubarak.
>
>For those who do not know what it is, Its a Prayer in memory of Ishmael son
of 
>Abraham, when he attempted to sacrifice his beloved son on request by god.
>
>Muslim's celebrate it with a sacrifice of a goat every year.
>
>I belive Christians &  Jew's belive in the same.
>  
>
Well, I'm an atheist, so... happy Asterisk!

For those who do not know what it is, Its a program in memory son of 
root, when he attempted to kill -15 his beloved AGI processes on request 
of BOFH.

Admins celebrate it with a sudo reboot every year (well, every week for 
windows admins).

I believe Asterisk & SER users believe in the same :)

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Re: [Asterisk-Users] codecs order and so on

2006-01-10 Thread Moises Silva
Doing in the console "show translation" i can see that it seems not be
possible to translate from any to g729 codec, or from g729 to any. So,
let me try to find a reason for this.

 When you have first allow=g729  (preferred codec)
all the calls to pstn providers work because the phones and asterisk
agree to use g729, so no codec translation is done.
all the calls to and from fxo fails because no translation can be made
from ULAW to g729, and from g729 (phones) to ulaw.
then asterisk is not smart enough to realize that can ask the phones
to use ulaw (i assume the phones support ulaw) to not use translation
to call the fxo???

 When you have first allow=ulaw (prefered codec)
all the calls to and from fxo works because the prefered codec is
ulaw, then from fxo to phones using ulaw, no codec translation is made
all the calls to pstn providers fails, again, because it seems
asterisk gives preference to ulaw codec (the first list codec) so, the
phones use ulaw, and is not possible to translate ulaw to g729 and
viceversa??

im interested in knowing the reason too, any guidelines?

regards

On 1/10/06, Olivier Taylor <[EMAIL PROTECTED]> wrote:
>
> The problem :
>
> an asterisk box with 2 fxo
>
> First fxo just receive calls from pstn (ulaw)
> Second fxo receive and send call to mobile network thru a sipbox(ulaw)
> Calls to pstn are sent to a pstn provider accepting only g729
> Internal calls doesn't care of codecs
> All Uas have g729 (g729 is then pass-thru when needed)
> All Uas have ulaw(of course)
> If I have in [general]
> disallow=all
> allow=g729
> allow=ulaw
>
> In this case:
>
> all calls to pstn providers works
> all calls to and from fxo fails because of : No translator path exists for
> 
>
> If I have in [general]
> disallow= all
> allow= ulaw
> allow= g729
>
> In this case:
>
> all calls to and from fxo works
> all calls to pstn providers fails because of : No translator path exists for
> 
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>
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>
>


--
"Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org";
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RE: [Asterisk-Users] Help with amportal: asterisk ended with exit status 127

2006-01-10 Thread Colin Anderson
Title: Message



from 
the command line, run asterisk -v. You will either get "Asterisk Ready" (good) 
or "Ouch...Broken Pipe" (bad). If it breaks, note the name of the last .so 
file it referenced before you got the "Ouch" and remove that file from 
/var/lib/asterisk/modules. Rinse and repeat until you get an "Asterisk Ready" 
prompt. Once you do, Ctrl-C (kill) it, then run ampportal start. Asterisk should 
run OK. 
 
Now 
you have to figure out why the .so modules you removed won't run. However, a lot 
of times they aren't important for you to use, so you can ignore them. For 
example, app_adsiprog.so is only relevant if you are running ADSI phones. If you 
aren't, and it's causing Asterisk to die, leave it out. 
 
One 
other thing comes to mind and that is you might be running safe_asterisk 
*already* from ampportal start which is invoked by default in 
/etc/rc.d/rc.local. Comment out the /usr/sbin/ampportal start in rc.local, 
reboot, THEN try the above stuff. If Asterisk runs right away, then it is a 
conflict with the entry in rc.local. You can comment it back in, reboot, then 
type in asterisk -r at the command line to get an Asterisk console. If not, you 
have to follow the troubleshooting procedure above in order to get it running. 

 
hth


  -Original Message-From: Ben Ferguson 
  [mailto:[EMAIL PROTECTED]Sent: Tuesday, January 10, 2006 1:48 
  PMTo: 'Asterisk Users Mailing List - Non-Commercial 
  Discussion'Subject: [Asterisk-Users] Help with amportal: asterisk 
  ended with exit status 127
  
  Greetings. I am trying to get AMP up and going on my Asterisk 
  server. I can access the admin pages on my asterisk server via a web browser. 
  I can add and edit things via the web browser and it edits the database 
  accordingly. Everything seems fine except when I try to run 'amportal start'. 
  Below is what I get (Plus tail /var/log/asterisk/full, but the tail of the 
  'full' log doesn't seem to list anything from trying to run amportal. It seems 
  to me that the last lines from the 'full' log are related to me stopping my 
  currently up and running asterisk so that I can try to run the amportal 
  command. Make sense?). Please help.
  Thanks,
  Ben F
  CirclePix
  
  [EMAIL PROTECTED] /usr/sbin/amportal start
  SETTING FILE PERMISSIONS
  Permissions OK
  STARTING ASTERISK
  Asterisk ended with exit status 127
  Asterisk died with code 127.
  Automatically restarting Asterisk.
  Asterisk ended with exit status 127
  Asterisk died with code 127.
  Automatically restarting Asterisk.
  mpg123: no process killed
  -
  Asterisk could not start!
  Use 'tail /var/log/asterisk/full' to find out why.
  -
  [EMAIL PROTECTED]
  [EMAIL PROTECTED] tail /var/log/asterisk/full
  Jan 10 09:47:01 VERBOSE[12657] logger.c: == Registered file 
  format pcm, extension(s) pcm|ulaw|ul|mu
  Jan 10 09:47:01 VERBOSE[12657] logger.c: == Manager registered 
  action DBGet
  Jan 10 09:47:01 VERBOSE[12657] logger.c: == Manager registered 
  action DBPut
  Jan 10 09:47:01 VERBOSE[12657] logger.c: == Parsing 
  '/etc/asterisk/enum.conf': Jan 10 09:47:01 VERBOSE[12657] logger.c: == Parsing 
  '/etc/asterisk/enum.conf': Found
  Jan 10 09:47:01 VERBOSE[12657] logger.c: Asterisk 
  Ready.
  Jan 10 09:47:09 VERBOSE[12657] logger.c: Asterisk Ready. 
  Beginning asterisk shutdown Jan 10 09:47:09 VERBOSE[12657] logger.c: 
  Executing last minute cleanups
  Jan 10 09:47:09 VERBOSE[12657] logger.c: == Destroying 
  musiconhold processes
  Jan 10 09:47:09 VERBOSE[12657] logger.c: Asterisk cleanly 
  ending (0). [EMAIL PROTECTED]
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[Asterisk-Users] Help with amportal: asterisk ended with exit status 127

2006-01-10 Thread Ben Ferguson
Title: Message




Greetings. I am trying to get AMP up and going on my Asterisk server. I can 
access the admin pages on my asterisk server via a web browser. I can add and 
edit things via the web browser and it edits the database accordingly. 
Everything seems fine except when I try to run 'amportal start'. Below is what I 
get (Plus tail /var/log/asterisk/full, but the tail of the 'full' log doesn't 
seem to list anything from trying to run amportal. It seems to me that the last 
lines from the 'full' log are related to me stopping my currently up and running 
asterisk so that I can try to run the amportal command. Make sense?). Please 
help.
Thanks,
Ben F
CirclePix
[EMAIL PROTECTED] /usr/sbin/amportal start
SETTING FILE PERMISSIONS
Permissions OK
STARTING ASTERISK
Asterisk ended with exit status 127
Asterisk died with code 127.
Automatically restarting Asterisk.
Asterisk ended with exit status 127
Asterisk died with code 127.
Automatically restarting Asterisk.
mpg123: no process killed
-
Asterisk could not start!
Use 'tail /var/log/asterisk/full' to find out why.
-
[EMAIL PROTECTED]
[EMAIL PROTECTED] tail /var/log/asterisk/full
Jan 10 09:47:01 VERBOSE[12657] logger.c: == Registered file format pcm, 
extension(s) pcm|ulaw|ul|mu
Jan 10 09:47:01 VERBOSE[12657] logger.c: == Manager registered action 
DBGet
Jan 10 09:47:01 VERBOSE[12657] logger.c: == Manager registered action 
DBPut
Jan 10 09:47:01 VERBOSE[12657] logger.c: == Parsing 
'/etc/asterisk/enum.conf': Jan 10 09:47:01 VERBOSE[12657] logger.c: == Parsing 
'/etc/asterisk/enum.conf': Found
Jan 10 09:47:01 VERBOSE[12657] logger.c: Asterisk Ready.
Jan 10 09:47:09 VERBOSE[12657] logger.c: Asterisk Ready. Beginning asterisk 
shutdown Jan 10 09:47:09 VERBOSE[12657] logger.c: Executing last minute 
cleanups
Jan 10 09:47:09 VERBOSE[12657] logger.c: == Destroying musiconhold 
processes
Jan 10 09:47:09 VERBOSE[12657] logger.c: Asterisk cleanly ending (0).
[EMAIL PROTECTED]
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[Asterisk-Users] TE405p -- loopback for the phone company?

2006-01-10 Thread Eric Lyons

Feeling once again like an idiot, I come to the list for help...

While my carrier (MCI vi PRI, fwiw) was trying to diagnose a problem, they asked me to put my interface in loopback -- and I 
couldn't figure out what they meant or how to do it.  I've plugged a loopback *connector* into my 405p and tested it internally, but 
they wanted something that, apparently, looped back _their_ signal.


What the?

I got zttool running and selected "loop" on the interface, but it didn't seem to do what they wanted (nor could I tell that it did 
anything at all).  Many googles for zaptel and loop didn't turn up anything useful.


Eric. 


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RE: [Asterisk-Users] The second edition of my Asterisk book is nowavailable

2006-01-10 Thread Doug G
Agreed, the first book kind of looked the WIKI in print..  



-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver
Sent: Tuesday, January 10, 2006 3:44 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Cc: asterisk-biz@lists.digium.com
Subject: Re: [Asterisk-Users] The second edition of my Asterisk book is 
nowavailable

[EMAIL PROTECTED] a écrit :

>The second edition of my book "VoIP Telephony with Asterisk" is now in
>print and available. You can find out more about it at our web site
>http://www.signate.com/products.php
>
>This book is written for beginners. It will make it easier for you to get
>started. The second edition is reorganized and expanded.*
>  
>
Is there a TOC somewhere, so I can compare with the 1st version (which I 
bought)?

The 1st version certainly gave me some insights, but to be honest I did 
find it was looking more like a patchwork of recipies rather than 
following a proper "teaching line".

Cheers,
Jean-Michel.

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Re: [Asterisk-Users] Help with amportal: asterisk ended with exit status 127

2006-01-10 Thread Moises Silva
hum... 127 is command not found, right??

On 1/10/06, Ben Ferguson <[EMAIL PROTECTED]> wrote:
>
>
>
> Greetings. I am trying to get AMP up and going on my Asterisk server. I can
> access the admin pages on my asterisk server via a web browser. I can add
> and edit things via the web browser and it edits the database accordingly.
> Everything seems fine except when I try to run 'amportal start'. Below is
> what I get (Plus tail /var/log/asterisk/full, but the tail of the 'full' log
> doesn't seem to list anything from trying to run amportal. It seems to me
> that the last lines from the 'full' log are related to me stopping my
> currently up and running asterisk so that I can try to run the amportal
> command. Make sense?). Please help.
>
> Thanks,
>
> Ben F
>
> CirclePix
>
> 
>
> [EMAIL PROTECTED] /usr/sbin/amportal start
>
> SETTING FILE PERMISSIONS
>
> Permissions OK
>
> STARTING ASTERISK
>
> Asterisk ended with exit status 127
>
> Asterisk died with code 127.
>
> Automatically restarting Asterisk.
>
> Asterisk ended with exit status 127
>
> Asterisk died with code 127.
>
> Automatically restarting Asterisk.
>
> mpg123: no process killed
>
> -
>
> Asterisk could not start!
>
> Use 'tail /var/log/asterisk/full' to find out why.
>
> -
>
> [EMAIL PROTECTED]
>
> [EMAIL PROTECTED] tail /var/log/asterisk/full
>
> Jan 10 09:47:01 VERBOSE[12657] logger.c: == Registered file format pcm,
> extension(s) pcm|ulaw|ul|mu
>
> Jan 10 09:47:01 VERBOSE[12657] logger.c: == Manager registered action DBGet
>
> Jan 10 09:47:01 VERBOSE[12657] logger.c: == Manager registered action DBPut
>
> Jan 10 09:47:01 VERBOSE[12657] logger.c: == Parsing
> '/etc/asterisk/enum.conf': Jan 10 09:47:01 VERBOSE[12657] logger.c: ==
> Parsing '/etc/asterisk/enum.conf': Found
>
> Jan 10 09:47:01 VERBOSE[12657] logger.c: Asterisk Ready.
>
> Jan 10 09:47:09 VERBOSE[12657] logger.c: Asterisk Ready. Beginning asterisk
> shutdown Jan 10 09:47:09 VERBOSE[12657] logger.c: Executing last minute
> cleanups
>
> Jan 10 09:47:09 VERBOSE[12657] logger.c: == Destroying musiconhold processes
>
> Jan 10 09:47:09 VERBOSE[12657] logger.c: Asterisk cleanly ending (0).
> [EMAIL PROTECTED]
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Re: [Asterisk-Users] 32 E1's in one Asterisk 'box'

2006-01-10 Thread [EMAIL PROTECTED]






  Contemplating about thirtytwo E1 lines?
Perhaps it's time to start thinking about asterisk support for PDH or
SDH... 
In this case, two 34Mb (OC1) boards would do nicely.

  

I know of boards that will enable me to mount 32 E1's into Asterisk
assuming I write the drivers for them (DSP based cPCI) - that is not
the real problem. The issue is that I need to split the traffic over
several PC's to reduce the loss if one goes down - redundancy etc. But,
I still need this to act as if it was 'one box'. An ideal design seen
from my point would be 4 PC's with 8 E1's each and when 2 extra PC's
just for PABX call control etc (one standby for the other).

I was kond-of hoping this had been solved, but I assume I am wrong???

Didn't Digium work on an OC1/E3 board ? Plenty of framer's on a chip so
it is fully possible to make decent priced E3/PCI etc.

Jan


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[Asterisk-Users] testmail

2006-01-10 Thread T A
I have not received any mail since 14:06 CET, so I am just testing if this
goes through. 

I had some mailserver trouble at the time stated earlier, so I am wondering
if something happend during this time. Something like my mail adress
returning a lot of bounces, and therefore beeing blocked?

Regards,
TA

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Re: [Asterisk-Users] Second edition of my * book has been released

2006-01-10 Thread Randy Williams

Greetings All,

I have found that Paul's book is just right for rounding out the edges 
when getting started.  I managed to temporarily migrate our T-1 Asterisk 
system to a Analog asterisk system on information in Paul's book alone.  
Nicely done and a neat bit of help in a pinch.


Just my $0.02 USD you understand. :)

RandyW

Paul Mahler wrote:


Hi Greg,

My book is a good place for a beginner to get started. I also find it to be
useful as a reference for Asterisk. It's not an advanced book, there are
advanced features it doesn't cover, for example AGI or the management
interface. 


It should be very helpful for your customers. It should be helpful for a
beginning to intermediate administrator. I still frequently refer to it
myself when I'm having a senior moment. :) 


There isn't anything in the book that would make it less useful for the CVS
or stable branches. 


The O'Reilly book is excellent. I think my book complements the O'Reilly
book. If I were just starting I would buy both. I think my book may be a bit
more useful as a reference. I think I cover a bit more beginner's territory.


Hope This Helps,

Paul

 


-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Monday, January 09, 2006 9:10 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Second edition of my * book has been
released

How does it compare with the O'Rielly book?

Does it include information on CVS, or primarily on stable?

Can it be provided to customers, or is it more sysadmin oriented?

Regards,
Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
Mahler
Sent: Thursday, January 05, 2006 9:45 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Second edition of my * book has been released

The second edition of my Asterisk book "VoIP Telephony with Asterisk" is
now in print. It's reorganized and expanded.

TKS

Paul Mahler


Paul Mahler
[EMAIL PROTECTED]

www.signate.com


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No virus found in this incoming message.
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Re: [Asterisk-Users] MTU and Voice Delay (latency??)

2006-01-10 Thread pdhales
Same experience here!

I had a tech in Perth call my desk phone in Melbourne, which was forwarded
to my mobile.
He was stunned how good it sounded, and thought I was in Perth too, on a
desk phone.

PaulH

- Original Message - 
From: "Matt Riddell (IT)" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Tuesday, January 10, 2006 11:22 PM
Subject: Re: [Asterisk-Users] MTU and Voice Delay (latency??)


> [EMAIL PROTECTED] wrote:
> > We had phones in Perth hooked up to an Asterisk box in Melbourne, and
the call was fine - so I know it can be done.
>
> I'm currently in Italy and I've had a few conversations that lasted a
> few minutes before the person at the other end realised I was in Italy
> :) (from New Zealand)
>
> -- 
> Cheers,
>
> Matt Riddell
> ___
>
> http://www.sineapps.com/news.php (Daily Asterisk News - html)
> http://freevoip.gedameurope.com (Free Asterisk Voip Community)
> http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
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Re: [Asterisk-Users] The second edition of my Asterisk book is now available

2006-01-10 Thread Chris Tooley

> >
> Is there a TOC somewhere, so I can compare with the 1st version (which I 
> bought)?
> 
http://www.osoft.com/store/productdetails.php?pid=151


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Re: [Asterisk-Users] pattern mach doubt

2006-01-10 Thread C F
It should work.

On 1/10/06, Dov Bigio <[EMAIL PROTECTED]> wrote:
>
> Hi ALL,
>
> Is it possible to use symbols # and * in the dialplan for pattern matching?
>
> I am doing a "follow me" dial plan, and wanted that my users could dial
> everything in one shot.
>
> But, exten => 888*XXX*XXX,1,NoOp(Follow Me from XXX to XXX)
> doesn't seem to work...

Did you try:
exten => _888*XXX*XXX,1,NoOp(Follow Me from XXX to XXX)

>
> Thank you
> Dov
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Re: [Asterisk-Users] Still an open Seat in London for Next Weeks Signate intro to Asterisk Course

2006-01-10 Thread pdhales
I would love to be there, but it's just too far to drive.

regards,

PaulH

- Original Message - 
From: <[EMAIL PROTECTED]>
To: ; 
Sent: Wednesday, January 11, 2006 7:24 AM
Subject: [Asterisk-Users] Still an open Seat in London for Next Weeks
Signate intro to Asterisk Course


> We still have a seat open in our Asterisk training course next week in
> London. You can find more information at our Web site, www.signate.com
>
> I'm going to be teaching the class.
>
> Paul
>
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[Asterisk-Users] Distorted indication tones after 1.0.9 -> 1.2.1 upgrade

2006-01-10 Thread Ola Lidholm
Hi all,

I have just upgraded to asterisk 1.2.1 from 1.0.9.

Almost everything worked right away.

However, the indication tones are now distorted. They sort of sound as
they should, but the sound quality is really crappy and "oversteared".

Has anyone else noticed that, and have a solution?

I am using swedish (se) indications.

--
/Ola

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[Asterisk-Users] Help with amportal: asterisk ended with exit status 127

2006-01-10 Thread Ben Ferguson
Title: Message




Greetings. I am trying to get AMP up and going on my Asterisk 
server. I can access the admin pages on my asterisk server via a web browser. I 
can add and edit things via the web browser and it edits the database 
accordingly. Everything seems fine except when I try to run 'amportal start'. 
Below is what I get (Plus tail /var/log/asterisk/full, but the tail of the 
'full' log doesn't seem to list anything from trying to run amportal. It seems 
to me that the last lines from the 'full' log are related to me stopping my 
currently up and running asterisk so that I can try to run the amportal command. 
Make sense?). Please help.
Thanks,
Ben F
CirclePix

[EMAIL PROTECTED] /usr/sbin/amportal start
SETTING FILE PERMISSIONS
Permissions OK
STARTING ASTERISK
Asterisk ended with exit status 127
Asterisk died with code 127.
Automatically restarting Asterisk.
Asterisk ended with exit status 127
Asterisk died with code 127.
Automatically restarting Asterisk.
mpg123: no process killed
-
Asterisk could not start!
Use 'tail /var/log/asterisk/full' to find out why.
-
[EMAIL PROTECTED]
[EMAIL PROTECTED] tail /var/log/asterisk/full
Jan 10 09:47:01 VERBOSE[12657] logger.c: == Registered file 
format pcm, extension(s) pcm|ulaw|ul|mu
Jan 10 09:47:01 VERBOSE[12657] logger.c: == Manager registered 
action DBGet
Jan 10 09:47:01 VERBOSE[12657] logger.c: == Manager registered 
action DBPut
Jan 10 09:47:01 VERBOSE[12657] logger.c: == Parsing 
'/etc/asterisk/enum.conf': Jan 10 09:47:01 VERBOSE[12657] logger.c: == Parsing 
'/etc/asterisk/enum.conf': Found
Jan 10 09:47:01 VERBOSE[12657] logger.c: Asterisk 
Ready.
Jan 10 09:47:09 VERBOSE[12657] logger.c: Asterisk Ready. 
Beginning asterisk shutdown Jan 10 09:47:09 VERBOSE[12657] logger.c: 
Executing last minute cleanups
Jan 10 09:47:09 VERBOSE[12657] logger.c: == Destroying 
musiconhold processes
Jan 10 09:47:09 VERBOSE[12657] logger.c: Asterisk cleanly ending 
(0). [EMAIL PROTECTED]
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Re: [Asterisk-Users] The second edition of my Asterisk book is now available

2006-01-10 Thread Jean-Michel Hiver

[EMAIL PROTECTED] a écrit :


The second edition of my book "VoIP Telephony with Asterisk" is now in
print and available. You can find out more about it at our web site
http://www.signate.com/products.php

This book is written for beginners. It will make it easier for you to get
started. The second edition is reorganized and expanded.*
 

Is there a TOC somewhere, so I can compare with the 1st version (which I 
bought)?


The 1st version certainly gave me some insights, but to be honest I did 
find it was looking more like a patchwork of recipies rather than 
following a proper "teaching line".


Cheers,
Jean-Michel.

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Re: [Asterisk-Users] 32 E1's in one Asterisk 'box'

2006-01-10 Thread Hans Witvliet
On Tue, 2006-01-10 at 12:03 +0100, [EMAIL PROTECTED] wrote:
> hi,
> 
> My apologies for repeating this question, but I hoped re-frasing it 
> might help.
> 
> I would like to assemble an PABX larger than what you possible can put 
> inside one Asterisk box. What is the best way to do this? Can it be done 
> at all with Asterisk? Any ideas or hints would be apreaciated.
> 
> jvb
> ___
Contemplating about thirtytwo E1 lines?
Perhaps it's time to start thinking about asterisk support for PDH or
SDH... 
In this case, two 34Mb (OC1) boards would do nicely.

Hans
-- 
pgp-id: 926EBB12
pgp-fingerprint: BE97 1CBF FAC4 236C 4A73  F76E EDFC D032 926E BB12
Registered linux user: 75761 (http://counter.li.org)
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Re: [Asterisk-Users] GrandSTream 488/Asterisk

2006-01-10 Thread Philip H W Schroth
Op dinsdag 10 januari 2006 21:27, schreef Jean-Michel Hiver:
I have it working with [EMAIL PROTECTED] Just give the fxo port an internal 
extension. Dial that extension and you are on the pstn..

Philip
> Goran Donev a écrit :
> >Has anyone tested a grandstream 488 FXO gateway on an Asterisk machine? I
> >read that the 488 has a FXO port on it, can I use the grandstream 488 to
> >pass traffic to the pstn from Asterisk.
>
> I didn't have much success with grandstream, although I got it to work
> with SIPURA 3k after some time fiddling around.
>
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RE: [Asterisk-Users] Recommendations on a WiFi phone for *?

2006-01-10 Thread Paul Mahler
I would be MUCH more tempted to use an IAXy or SIP adaptor and a cordless
phone. It will be less expensive and it will likely work better. 

Paul

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Joash Herbrink
> Sent: Monday, January 09, 2006 11:53 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Recommendations on a WiFi phone for *?
> 
> The zyxel p2000W
> Works fine, good batt. Live.
> Decent sound quality.
> 
> All in all a good product for about 150 euro's
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Philip
> Edelbrock
> Sent: Tuesday, January 10, 2006 2:45 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Recommendations on a WiFi phone for *?
> 
> 
> We're getting our feet more and more wet with VOIP at work.  We want to
> experiment with a good wireless (as in WiFi) phone.  What would be a
> good phone to impress my boss with?
> 
> I'm personally drooling over the UTStarcom F3000, but compatibility and
> shipping ETA info is a bit sketchy.
> 
> 
> Phil
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> 
> 
> --
> No virus found in this incoming message.
> Checked by AVG Free Edition.
> Version: 7.1.371 / Virus Database: 267.14.15/223 - Release Date: 1/6/2006


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RE: [Asterisk-Users] Second edition of my * book has been released

2006-01-10 Thread Paul Mahler
Hi Greg,

My book is a good place for a beginner to get started. I also find it to be
useful as a reference for Asterisk. It's not an advanced book, there are
advanced features it doesn't cover, for example AGI or the management
interface. 

It should be very helpful for your customers. It should be helpful for a
beginning to intermediate administrator. I still frequently refer to it
myself when I'm having a senior moment. :) 

There isn't anything in the book that would make it less useful for the CVS
or stable branches. 

The O'Reilly book is excellent. I think my book complements the O'Reilly
book. If I were just starting I would buy both. I think my book may be a bit
more useful as a reference. I think I cover a bit more beginner's territory.


Hope This Helps,

Paul

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
> Sent: Monday, January 09, 2006 9:10 PM
> To: asterisk-users@lists.digium.com
> Subject: RE: [Asterisk-Users] Second edition of my * book has been
> released
> 
> How does it compare with the O'Rielly book?
> 
> Does it include information on CVS, or primarily on stable?
> 
> Can it be provided to customers, or is it more sysadmin oriented?
> 
> Regards,
> Greg
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Paul
> Mahler
> Sent: Thursday, January 05, 2006 9:45 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [Asterisk-Users] Second edition of my * book has been released
> 
> The second edition of my Asterisk book "VoIP Telephony with Asterisk" is
> now in print. It's reorganized and expanded.
> 
> TKS
> 
> Paul Mahler
> 
> 
> Paul Mahler
> [EMAIL PROTECTED]
> 
> www.signate.com
> 
> 
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> 
> 
> --
> No virus found in this incoming message.
> Checked by AVG Free Edition.
> Version: 7.1.371 / Virus Database: 267.14.15/223 - Release Date: 1/6/2006


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Re: [Asterisk-Users] Eid Mubarak

2006-01-10 Thread Jean-Michel Hiver

Rehan AllahWala a écrit :


Dear All,

For those who celebrate Eid.

I would like to wish you a very Happy Eid Mubarak.

For those who do not know what it is, Its a Prayer in memory of Ishmael son of 
Abraham, when he attempted to sacrifice his beloved son on request by god.


Muslim's celebrate it with a sacrifice of a goat every year.

I belive Christians &  Jew's belive in the same.
 


Well, I'm an atheist, so... happy Asterisk!

For those who do not know what it is, Its a program in memory son of 
root, when he attempted to kill -15 his beloved AGI processes on request 
of BOFH.


Admins celebrate it with a sudo reboot every year (well, every week for 
windows admins).


I believe Asterisk & SER users believe in the same :)

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[Asterisk-Users] The second edition of my Asterisk book is now available

2006-01-10 Thread paul . mahler
The second edition of my book "VoIP Telephony with Asterisk" is now in
print and available. You can find out more about it at our web site
http://www.signate.com/products.php

This book is written for beginners. It will make it easier for you to get
started. The second edition is reorganized and expanded.

Thanks,

Paul

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Re: [Asterisk-Users] GrandSTream 488/Asterisk

2006-01-10 Thread Jean-Michel Hiver

Goran Donev a écrit :


Has anyone tested a grandstream 488 FXO gateway on an Asterisk machine? I
read that the 488 has a FXO port on it, can I use the grandstream 488 to
pass traffic to the pstn from Asterisk.
 

I didn't have much success with grandstream, although I got it to work 
with SIPURA 3k after some time fiddling around.


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[Asterisk-Users] Still an open Seat in London for Next Weeks Signate intro to Asterisk Course

2006-01-10 Thread paul . mahler
We still have a seat open in our Asterisk training course next week in
London. You can find more information at our Web site, www.signate.com

I'm going to be teaching the class.

Paul

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[Asterisk-Users] Eid Mubarak

2006-01-10 Thread Rehan AllahWala
Dear All,

For those who celebrate Eid.

I would like to wish you a very Happy Eid Mubarak.

For those who do not know what it is, Its a Prayer in memory of Ishmael son of 
Abraham, when he attempted to sacrifice his beloved son on request by god.

Muslim's celebrate it with a sacrifice of a goat every year.

I belive Christians &  Jew's belive in the same.

Peace and Harmony to all.

Rehan


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[Asterisk-Users] Sipura SPA-2100 / 3000 provisioning .xml examples / xml variable list

2006-01-10 Thread Vahan Yerkanian

Hi,

I'm looking for a full list of xml provisioning variables of the 
SPA-2100/3000. Currently the Sipura website has example XMLs only for 
the SPA-841 [1] and  SPA-941 [2].


I'm mostly interested in the CallerID type selector variables and 
whatever variables control the PSTN<->VoIP settings. Sipura 
Configuration website form field names are numeral only. :(


[1] http://www.sipura.com/support/spa841faq/sample-841.xml
[2] http://www.sipura.com/support/spa941faq/sample-941.xml

Best regards,
Vahan

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RE: [Asterisk-Users] Asterisk voicemail support

2006-01-10 Thread S McGowan



That's because DELETE is a reserved 
word. The queries asterisk sends need to have ` surrounding the name, not ' or 
".
 
I had the same problem but my 
employers at the time didn't want to delve into the code, we just used the 
options column

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of 
  AislingSent: Tuesday, January 10, 2006 12:57 PMTo: 
  asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk 
  voicemail support
  
  
  Hi,
   
  I was wondering if anyone has had 
  a problem adding the ‘delete’ field to the voicemail_users table. I have no problems adding other 
  fields e.g.
   
  alter table 
  voicemail_users add column hidefromdir varchar(3) NOT NULL 
  default ‘no’;
   
  However when I do 
  
   
  alter table 
  voicemail_users add column delete varchar(3) NOT NULL default 
  ‘no’;
   
  I get a message telling me that I 
  have an error in my MySQL syntax…..Is this because the ‘delete’ word I s a 
  reserved word and if so is this something others have 
  experienced?
   
  Many 
  thanks,
  Aisling.---Legal 
  Disclaimer--- The above electronic mail 
  transmission is confidential and intended only for the person to whom it is 
  addressed. Its contents may be protected by legal and/or professional 
  privilege. Should it be received by you in error please contact the sender at 
  the above quoted email address. Any unauthorised form of reproduction of this 
  message is strictly prohibited. The Institute does not guarantee the security 
  of any information electronically transmitted and is not liable if the 
  information contained in this communication is not a proper and complete 
  record of the message as transmitted by the sender nor for any delay in its 
  receipt. 
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Re: [Asterisk-Users] controlling SIP subscriptions from SNOM phones

2006-01-10 Thread Bob Goddard
On Monday 09 Jan 2006 13:46, Sven Fischer (support) wrote:
> On Saturday 07 January 2006 02:30, Philipp von Klitzing wrote:
> > Hi!
> >
> > > Now, one user, not the receptionist, has gone in and set his personal
> > > numbers to these function keys thinking that DESTINATION meant setting
> > > a number to dial out. So now I have a ton of SIP SUBSCRIBE messages for
> > > his numbers.
> >
> > Indeed this situation is not ideal. The first thing to do in my opinion
> > is ask SNOM to provide a new type of DESTINATION option that does not
> > issue subscribes.
>
> This is already available with firmware release 5 for snom320/360. This new
> type is named "speed dial".

I guess this means it will not be available for the 190/200/220's?
I assume that the software development for these phones has ceased?


B

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Re: [Asterisk-Users] Re: web sip client

2006-01-10 Thread Rehan AllahWala
The guy who wrote it is

Babar Shafiq  I have seen but not used
> http://www.silicontechnix.com/webtelefone/start.html
> 
> -- 
> -- 
> Steven
> 
> May you have the peace and freedom that come from abandoning all hope
> of having a better past. ----  ---  - - -   -- -  
> -   --  - - - --- - --   - - --- - - -- -  -- --   -   --
> "Miguel" <[EMAIL PROTECTED]> wrote in message
> news:[EMAIL PROTECTED] > Hi, does anyone knows a good
> (comercial/oss) web client for asterisk? i mean similar to a softphone
> but using some kind of web > interface, ideally a would create all the
> user/pass in asterisk, the customer logs in using a web form and he
> can make calls using > the web interface, something similar to
> dial-pad, i dont mind to pay for this > thanks >
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Super Technologies Inc., Pensacola, Florida
http://www.SuperTec.com - Technologies from tomorrow, Today!

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[Asterisk-Users] GrandSTream 488/Asterisk

2006-01-10 Thread Goran Donev
Has anyone tested a grandstream 488 FXO gateway on an Asterisk machine? I
read that the 488 has a FXO port on it, can I use the grandstream 488 to
pass traffic to the pstn from Asterisk.

I would use this at home to pass traffic into a foreign country's PSTN over
the internet. 

Thanks.


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Re: [Asterisk-Users] Recommendations on a WiFi phone for *?

2006-01-10 Thread Michiel van Baak
On 13:35, Tue 10 Jan 06, O'Connor, Jonathan wrote:
> I've been looking around to see where I could get one, would solve a
> problem we have quite nicely.
> 
> Where did you get yours if I may ask?

And how much are they if I may ask ?
-- 
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] Re: web sip client

2006-01-10 Thread Steven
I have seen but not used http://www.silicontechnix.com/webtelefone/start.html

-- 
-- 
Steven

May you have the peace and freedom that come from abandoning all hope of having 
a better past.
----  ---  - - -   -- -   -   --  - - - --- - --   - - 
--- - - -- -  -- --   -   --
"Miguel" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
> Hi, does anyone knows a good (comercial/oss) web client for asterisk? i mean 
> similar to a softphone but using some kind of web 
> interface, ideally a would create all the user/pass in asterisk, the customer 
> logs in using a web form and he can make calls using 
> the web interface, something similar to dial-pad, i dont mind to pay for this
> thanks
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[Asterisk-Users] outboundproxy issue

2006-01-10 Thread mcquaid mcquaid
Hello, new to asterisk and trying to set it up to work with my voip provider (vbuzzer.com).  I am behind a firewall that I don't have access to, to open ports etc.  Before using asterisk, I tried vbuzzer's windows client, and linphone and twinklephone which all worked without having to enable nat or stun.  However I did have to enter the outboundproxy server to get them to function.  Not sure if it's an issue but my voip provider uses port 80 for sip instead of 5060.
In asterisk, I can make outgoing calls through my voip provider to pstn lines, audio works both ways.  But calling in from my land line to the asterisk box via vbuzzer, I get no audio either way.  The local sip client rings and when I answer the call,  I see asterisk sending/receiving rtp packets. I couldn't find much information on asterisk's outboundproxy and outboundproxyport variables.  They were in chan_sip2 last year and then merged with chan_sip.  At that time, there were a glofal var, but now I think they can be a peer.  I then tried just having the asterisk server answer incoming calls from vbuzzer, and I see it saying it's playing monkeys, but no audio.  Again, if I dial from within to the asterisk box via a local sip client I get audio.  I might be on the wrong track with the outboundproxy, but since I'm not setting nat or stun in the other sip clients and they can make and receive calls, I can't see what else it could be.
I also read about siproxd, and it said in it's docs that (at that time) asterisk didn't support outboundproxy and siproxd could be used as a transparent proxy.  Could siproxd be used behind the firewall as I am?  I'm running asterisk on my local box not on the firewall I'm behind.  If someone has experience with siproxd, I'd like to give it a try, but I don't see how to tie it in with asterisk and a voip provider.  

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RE: [Asterisk-Users] Pri Gateway Hardware

2006-01-10 Thread Carlos Alperin
Low level requeriment, just you transfer everything using level 2. So you
don't need to the overhead to have Asterisk running to route that traffic.

Carlos Alperin

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel
Hiver
Sent: Tuesday, January 10, 2006 1:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Pri Gateway Hardware

Alexander Lopez a écrit :

>TDMoE is stable and stale, There is no more development planed or needed as
it only opens up a pipe between two ethernet points using Layer 2.
>  
>
OK... What would be in the advantage in running TDMoE rather than using 
IAX or SIP?

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RE: [Asterisk-Users] Recommendations on a WiFi phone for *?

2006-01-10 Thread O'Connor, Jonathan
I've been looking around to see where I could get one, would solve a
problem we have quite nicely.

Where did you get yours if I may ask?

-Jonathan


 
Jonathan O'Connor
Senior System Administrator
Inoveris LLC
Direct Line (614) 791-3742
Fax (614) 791-3748
Helpdesk 866-456-1566
 
 
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Benjamin Lawetz
> Sent: Tuesday, January 10, 2006 1:03 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Recommendations on a WiFi phone for *?
> 
> Actually got my hands on one, it's not that bad size wise. 
> About the size of a big smartphone
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Jean-Michel Hiver
> Sent: January 10, 2006 11:36 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Recommendations on a WiFi phone for *?
> 
> 
> >
> > 
> http://www.paesys.com/en/GSM_Wi-Fi_phone_for_SIP_voice_and_data_GTEK_P
> > WG500.htm
> >
> 
> Yeah I've seen that one... to bad it looks like such a brick :(
> 
> Cheers,
> Jean-Michel.
> 
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> 
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Re: [Asterisk-Users] hangup detection

2006-01-10 Thread steve


On Tue, 10 Jan 2006, [EMAIL PROTECTED] wrote:

> Thanks for your suggestion Steve.
> I have done as you advised and set  busypattern=300,200 to match the sample
> I recorded.
> This hasn't worked though, asterisk doesn't seem to detect the busy signal.
> Does asterisk require a the signal to be in a certain power range?  The
> signal I get
> is very quiet.
> Thanks for your help
> Regards
> Jonathan

Yeah - it needs to be reasonably loud to be detected.  Too bad.

Steve

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Re: [Asterisk-Users] Pri Gateway Hardware

2006-01-10 Thread Jean-Michel Hiver

Alexander Lopez a écrit :


TDMoE is stable and stale, There is no more development planed or needed as it 
only opens up a pipe between two ethernet points using Layer 2.
 

OK... What would be in the advantage in running TDMoE rather than using 
IAX or SIP?


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Re: [Asterisk-Users] CHAN_CAPI problem

2006-01-10 Thread Armin Schindler
On Tue, 10 Jan 2006 [EMAIL PROTECTED] wrote:
> Thank you.
> I already downloaded and installed it (they are dated 07-01-2006, version
> 0.6.3, correct ?)

Yes.

> I maked clean, make and make install.
> 
> Nothing changed, dial out perfect, dial in: (capi debug on)
> 
> asteriskge03*CLI> capi info
> Contr1: 2 B channels total, 2 B channels free.
> asteriskge03*CLI>
> asteriskge03*CLI>
> -- CONNECT_IND
> (PLCI=0x101,DID=104695467,CID=108680550,CIP=0x1,CONTROLLER=0x1)
>   == BRI1: Interface cleanup PLCI=0x101
> 
> BRI1 is the name of my interface
> 
> could it be a kernel issue ?? I am using SUSE Linux 10;

I don't think so. Please increase the the verbose level to 5
(set verbose 5) in addition to 'capi debug'.

Armin
 
> kernel :  2.6.13-15.7-smp
> 
> Andrea
> 
> 
> 
>
>  Armin Schindler   
>  <[EMAIL PROTECTED] 
>  >  To 
>  Sent by:  Asterisk Users Mailing List -   
>  asterisk-users-bo Non-Commercial Discussion   
>  [EMAIL PROTECTED]
>  m.com  cc 
>
>Subject 
>  10/01/2006 18.29  Re: [Asterisk-Users] CHAN_CAPI  
>problem 
>
>  Please respond to 
>   Asterisk Users   
>   Mailing List -   
>   Non-Commercial   
> Discussion 
>  <[EMAIL PROTECTED] 
>  ists.digium.com>  
>
>
> 
> 
> 
> 
> I suggest you use the newer chan_capi-cm (loadable from sourceforge.net).
> 
> Armin
> 
> On Tue, 10 Jan 2006 [EMAIL PROTECTED] wrote:
> > Hi all,
> > I installed asterisk stable cvs 1.2 and chan_capi 0.4.0 PRE1, with one
> AVM
> > Fritz Card ISDN connected to a Telecom NT1 Plus
> >
> > I configured asterisk via AMP.
> > No problem in making calls.
> > If I try to ring the ISDN Phone Number, I don't see anything on the
> > asterisk Console,
> > I I activate the capi debug , I see the ring on the capi channel.
> > If the context were wrong , I anyway should  see some line about this
> >
> > Why I cannot see anything on asterisk ,nor in the /var/log/asterisk/full
> ?
> >
> > here is my /etc/asterisk/capi.conf
> >
> > asteriskge03:/etc/asterisk # cat capi.conf
> > ;
> > ; CAPI config
> > ;
> > ;
> >
> > ; general section
> >
> > [general]
> > nationalprefix=0
> > internationalprefix=00
> > rxgain=0.8
> > txgain=0.8
> > ;ulaw=yes;set this, if you live in u-law world instead of a-law
> >
> > ; interface sections ...
> >
> > [BRI1]  ;this example interface gets name 'ISDN1' and may be any
> >  ;name not starting with 'g' or 'contr'.
> > ;ntmode=yes  ;if isdn card operates in nt mode, set this to yes
> > isdnmode=msn ;'MSN' (point-to-multipoint) or 'DID' (direct inward
> dial)
> >  ;when using NT-mode, 'DID' should be set in any case
> > incomingmsn=*;allow incoming calls to this list of MSNs/DIDs, * = any
> > ;defaultcid=123  ;set a default caller id to that interface for dial-out,
> >  ;this caller id will be used when dial option 'd' is
> set.
> > ;controller=0;ISDN4BSD default
> > ;controller=7;ISDN4BSD USB default
> > controller=1 ;capi controller number to use
> > group=1  ;dialout group
> > ;prefix=0;set a prefix to calling number on incoming calls
> > softdtmf=on  ;enable/disable software dtmf detection, recommended for
> > AVM cards
> > relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf
> > detection
> > accountcode= ;Asterisk accountcode to use in CDRs
> > context=from-pstn  ;context for incoming calls
> > holdtype=hold;when Asterisk puts the call on hold, ISDN HOLD will be
> > used. If
> >  ;set to 'local' (default value), no hold is done and
> > Asterisk may
> >  ;play MOH.
> > ;immediate=yes   ;DID: immediate start of pbx with extension 's' if no
> > digits were
> >  ; received on in

[Asterisk-Users] RE: Another cisco question

2006-01-10 Thread Brent Torrenga
Check out the Cisco SIP IP Phone Administrator Guide, Appendix D -
speed_line and speed_label

Do a google for Cisco SIP IP Phone Administrator Guide, easy peasy nice n
easy.

>Sorry about the unrelated questions about cisco phones, but does anyone
know how to set the second line up as a
> speed dial in the config file? 
>Or is that specifically a per-user basis setting?
>
>Aaron


Sincerely,

Brent A. Torrenga
[EMAIL PROTECTED]

Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771

219.836.8918x325 Voice
219.836.1138 Facsimile
www.torrenga.com
BEGIN:VCARD
VERSION:2.1
N:Torrenga;Brent;August;Mr.
FN:Brent August Torrenga
ORG:Torrenga Engineering, Inc.
TITLE:Designer
TEL;WORK;VOICE:(219) 836-8918
TEL;WORK;FAX:(219) 836-1138
ADR;WORK:;;907 Ridge Road;Munster;IN;46321-1771
LABEL;WORK;ENCODING=QUOTED-PRINTABLE:907 Ridge Road=0D=0AMunster, IN 46321-1771
URL;WORK:http://www.torrenga.com
EMAIL;PREF;INTERNET:[EMAIL PROTECTED]
REV:20040209T215756Z
END:VCARD
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Re: [Asterisk-Users] Re: using a Gigaset SX440isdn on a Diva 4BRI ?

2006-01-10 Thread Andreas Reich

Louis-David Mitterrand wrote:

It seems ptmp is required for NT mode to work with this phone. I tried
plugging the phone directly on the telco BRI and it worked fine too.


Phones are generally only for ptmp mode. Only PBXs use ptp mode. (Here 
in Germany the ptp mode is even called "PBX mode")



Andreas
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Re: [Asterisk-Users] Re: using a Gigaset SX440isdn on a Diva 4BRI ?

2006-01-10 Thread Louis-David Mitterrand
On Tue, Jan 10, 2006 at 06:52:43PM +0100, Louis-David Mitterrand wrote:
> On Tue, Jan 10, 2006 at 05:43:12PM +0100, Armin Schindler wrote:
> > On Tue, 10 Jan 2006, Louis-David Mitterrand wrote:
> > >   [C:4] 22:0188:202 - D-X(003) 02 01 7F
> > >   [C:4] 22:0189:202 - D-X(003) 02 01 7F
> > >   [C:4] 22:0190:202 - D-X(003) 02 01 7F
> > >   [C:4] 22:0191:201 - MDL-ERROR(G)
> > >   [C:4] 22:0191:202 - SIG-EVENT  0A
> > 
> > The diva card is sending (D-X), but does not receive anything (D-R). It 
> > looks like either the cross connection still isn't working or the protocol
> > is wrong.
> 
> OK, making some progress here: I removed "-u" (ptp mode) from the
> divactrl init string and now I can call in and out with my Gigaset
> handset!
> 
> It seems ptmp is required for NT mode to work with this phone. I tried
> plugging the phone directly on the telco BRI and it worked fine too.

For completeness: 100-ohm resistor adapters are required for the link
between phone and card to work (I have resistor-less cables with
separate adapters).

-- 
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[Asterisk-Users] Asterisk voicemail support

2006-01-10 Thread Aisling








Hi,

 

I was wondering if anyone has had a problem adding the ‘delete’
field to the voicemail_users table. I have no
problems adding other fields e.g.

 

alter table voicemail_users add column hidefromdir
varchar(3) NOT NULL default ‘no’;

 

However when I do 

 

alter table voicemail_users add column delete varchar(3)
NOT NULL default ‘no’;

 

I get a message telling me that I have an error in my MySQL
syntax…..Is this because the ‘delete’ word I s a reserved word
and if so is this something others have experienced?

 

Many thanks,

Aisling.




---Legal  Disclaimer---

The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.




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RE: [Asterisk-Users] Pri Gateway Hardware

2006-01-10 Thread Alexander Lopez
TDMoE is stable and stale, There is no more development planed or needed as it 
only opens up a pipe between two ethernet points using Layer 2.

 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Jean-Michel Hiver
> Sent: Tuesday, January 10, 2006 10:19 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Pri Gateway Hardware
> 
> Alexander Lopez a écrit :
> 
> > Check out Redfone.
> >  
> > http://www.red-fone.com
> >  
> > It converts PRI or T1 to TDMoE (TDM over Ethernet)
> 
> Isn't TDMoE considered "deprecated"?
> 
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RE: [Asterisk-Users] Recommendations on a WiFi phone for *?

2006-01-10 Thread Benjamin Lawetz
Actually got my hands on one, it's not that bad size wise. About the size of
a big smartphone

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel
Hiver
Sent: January 10, 2006 11:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Recommendations on a WiFi phone for *?


>
> http://www.paesys.com/en/GSM_Wi-Fi_phone_for_SIP_voice_and_data_GTEK_P
> WG500.htm
>

Yeah I've seen that one... to bad it looks like such a brick :(

Cheers,
Jean-Michel.

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Re: [Asterisk-Users] CHAN_CAPI problem

2006-01-10 Thread asterisk
Thank you.
I already downloaded and installed it (they are dated 07-01-2006, version
0.6.3, correct ?)
I maked clean, make and make install.

Nothing changed, dial out perfect, dial in: (capi debug on)

asteriskge03*CLI> capi info
Contr1: 2 B channels total, 2 B channels free.
asteriskge03*CLI>
asteriskge03*CLI>
-- CONNECT_IND
(PLCI=0x101,DID=104695467,CID=108680550,CIP=0x1,CONTROLLER=0x1)
  == BRI1: Interface cleanup PLCI=0x101

BRI1 is the name of my interface

could it be a kernel issue ?? I am using SUSE Linux 10;

kernel :  2.6.13-15.7-smp

Andrea



   
 Armin Schindler   
 <[EMAIL PROTECTED] 
 >  To 
 Sent by:  Asterisk Users Mailing List -   
 asterisk-users-bo Non-Commercial Discussion   
 [EMAIL PROTECTED]
 m.com  cc 
   
   Subject 
 10/01/2006 18.29  Re: [Asterisk-Users] CHAN_CAPI  
   problem 
   
 Please respond to 
  Asterisk Users   
  Mailing List -   
  Non-Commercial   
Discussion 
 <[EMAIL PROTECTED] 
 ists.digium.com>  
   
   




I suggest you use the newer chan_capi-cm (loadable from sourceforge.net).

Armin

On Tue, 10 Jan 2006 [EMAIL PROTECTED] wrote:
> Hi all,
> I installed asterisk stable cvs 1.2 and chan_capi 0.4.0 PRE1, with one
AVM
> Fritz Card ISDN connected to a Telecom NT1 Plus
>
> I configured asterisk via AMP.
> No problem in making calls.
> If I try to ring the ISDN Phone Number, I don't see anything on the
> asterisk Console,
> I I activate the capi debug , I see the ring on the capi channel.
> If the context were wrong , I anyway should  see some line about this
>
> Why I cannot see anything on asterisk ,nor in the /var/log/asterisk/full
?
>
> here is my /etc/asterisk/capi.conf
>
> asteriskge03:/etc/asterisk # cat capi.conf
> ;
> ; CAPI config
> ;
> ;
>
> ; general section
>
> [general]
> nationalprefix=0
> internationalprefix=00
> rxgain=0.8
> txgain=0.8
> ;ulaw=yes;set this, if you live in u-law world instead of a-law
>
> ; interface sections ...
>
> [BRI1]  ;this example interface gets name 'ISDN1' and may be any
>  ;name not starting with 'g' or 'contr'.
> ;ntmode=yes  ;if isdn card operates in nt mode, set this to yes
> isdnmode=msn ;'MSN' (point-to-multipoint) or 'DID' (direct inward
dial)
>  ;when using NT-mode, 'DID' should be set in any case
> incomingmsn=*;allow incoming calls to this list of MSNs/DIDs, * = any
> ;defaultcid=123  ;set a default caller id to that interface for dial-out,
>  ;this caller id will be used when dial option 'd' is
set.
> ;controller=0;ISDN4BSD default
> ;controller=7;ISDN4BSD USB default
> controller=1 ;capi controller number to use
> group=1  ;dialout group
> ;prefix=0;set a prefix to calling number on incoming calls
> softdtmf=on  ;enable/disable software dtmf detection, recommended for
> AVM cards
> relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf
> detection
> accountcode= ;Asterisk accountcode to use in CDRs
> context=from-pstn  ;context for incoming calls
> holdtype=hold;when Asterisk puts the call on hold, ISDN HOLD will be
> used. If
>  ;set to 'local' (default value), no hold is done and
> Asterisk may
>  ;play MOH.
> ;immediate=yes   ;DID: immediate start of pbx with extension 's' if no
> digits were
>  ; received on incoming call (no destination number
> yet)
>  ;MSN: start pbx on CONNECT_IND and don't wait for
> SETUP/SENDING-COMPLETE.
>  ; info like REDIRECTINGNUMBER may be lost, but this
is
> necessary for
>  ; drivers/pbx/telco which does not send SETUP or
> SENDING-COMPLETE.
> ;echosquelch=1   ;_VERY_PRIMITIVE_ echo suppression
> ;echocancel=yes  ;EICON 

[Asterisk-Users] Re: using a Gigaset SX440isdn on a Diva 4BRI ?

2006-01-10 Thread Louis-David Mitterrand
On Tue, Jan 10, 2006 at 05:43:12PM +0100, Armin Schindler wrote:
> On Tue, 10 Jan 2006, Louis-David Mitterrand wrote:
> > [C:4] 22:0188:202 - D-X(003) 02 01 7F
> > [C:4] 22:0189:202 - D-X(003) 02 01 7F
> > [C:4] 22:0190:202 - D-X(003) 02 01 7F
> > [C:4] 22:0191:201 - MDL-ERROR(G)
> > [C:4] 22:0191:202 - SIG-EVENT  0A
> 
> The diva card is sending (D-X), but does not receive anything (D-R). It 
> looks like either the cross connection still isn't working or the protocol
> is wrong.

OK, making some progress here: I removed "-u" (ptp mode) from the
divactrl init string and now I can call in and out with my Gigaset
handset!

It seems ptmp is required for NT mode to work with this phone. I tried
plugging the phone directly on the telco BRI and it worked fine too.


Thanks,

-- 
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[Asterisk-Users] pattern mach doubt

2006-01-10 Thread Dov Bigio



Hi ALL,
 
Is it possible to use symbols # and * in the 
dialplan for pattern matching?
 
I am doing a "follow me" dial plan, and wanted that 
my users could dial everything in one shot.
 
But, exten => 
888*XXX*XXX,1,NoOp(Follow Me from XXX to XXX) 
doesn't seem to work...
 
Thank you
Dov
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[Asterisk-Users] codecs order and so on

2006-01-10 Thread Olivier Taylor
Title: Message



The problem 
:
 
an asterisk box 
with 2 fxo

  First fxo just 
  receive calls from pstn (ulaw)
  Second fxo receive 
  and send call to mobile network thru a sipbox(ulaw)
  Calls to pstn are 
  sent to a pstn provider accepting only g729
  Internal calls 
  doesn't care of codecs
  All Uas have g729 
  (g729 is then pass-thru when needed)
  All Uas have 
  ulaw(of course)
If I have in 
[general]
disallow=all
allow=g729 
allow=ulaw 
 
In this 
case:
 
all calls to 
pstn providers works
all calls to 
and from fxo fails because of :  No translator path exists for 


If I have in [general]
disallow= all 
allow= ulaw  

allow= g729 
 
In this 
case:

all calls to 
and from fxo works
all calls to 
pstn providers fails because of : No translator path exists for 

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[Asterisk-Users] Help with amportal: asterisk ended with exit status 127

2006-01-10 Thread Ben Ferguson
Greetings.  I am trying to get AMP up and going on my Asterisk server.  I
can access the admin pages on my asterisk server via a web browser.  I can
add and edit things via the web browser and it edits the database
accordingly.  Everything seems fine except when I try to run 'amportal
start'.  Below is what I get (Plus tail /var/log/asterisk/full, but the tail
of the 'full' log doesn't seem to list anything from trying to run amportal.
It seems to me that the last lines from the 'full' log are related to me
stopping my currently up and running asterisk so that I can try to run the
amportal command.  Make sense?).  Please help.

Thanks,
Ben F
CirclePix

[EMAIL PROTECTED] /usr/sbin/amportal start

SETTING FILE PERMISSIONS
Permissions OK

STARTING ASTERISK
Asterisk ended with exit status 127
Asterisk died with code 127.
Automatically restarting Asterisk.
Asterisk ended with exit status 127
Asterisk died with code 127.
Automatically restarting Asterisk.
mpg123: no process killed

-
Asterisk could not start!
Use 'tail /var/log/asterisk/full' to find out why.
-
[EMAIL PROTECTED]

[EMAIL PROTECTED] tail /var/log/asterisk/full
Jan 10 09:47:01 VERBOSE[12657] logger.c:   == Registered file format pcm,
extension(s) pcm|ulaw|ul|mu
Jan 10 09:47:01 VERBOSE[12657] logger.c:   == Manager registered action
DBGet
Jan 10 09:47:01 VERBOSE[12657] logger.c:   == Manager registered action
DBPut
Jan 10 09:47:01 VERBOSE[12657] logger.c:   == Parsing
'/etc/asterisk/enum.conf': Jan 10 09:47:01 VERBOSE[12657] logger.c:   ==
Parsing '/etc/asterisk/enum.conf': Found
Jan 10 09:47:01 VERBOSE[12657] logger.c: Asterisk Ready.
Jan 10 09:47:09 VERBOSE[12657] logger.c: Asterisk Ready.
Beginning asterisk shutdown
Jan 10 09:47:09 VERBOSE[12657] logger.c: Executing last minute cleanups
Jan 10 09:47:09 VERBOSE[12657] logger.c:   == Destroying musiconhold
processes
Jan 10 09:47:09 VERBOSE[12657] logger.c: Asterisk cleanly ending (0).
[EMAIL PROTECTED]

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Re: [Asterisk-Users] voip-info: Asterisk record calls

2006-01-10 Thread Mojo with Horan & Company, LLC
Is there a Monitor application called in your dialplan?  It might have a 
basename parameter that configures this.  Or you could maybe call 
ChangeMonitor yourself but I don't know how to configure the timestamp 
portions of the filename.


Tim Litwiller wrote:

Mojo with Horan & Company, LLC wrote:


Hi Tim!

Wow, I didn't imagine that asterisk on different systems would use 
different date codes for the monitor filenames -- but aah isn't 
asterisk ;)


AAH builds Asterisk from source during the install. 


I just reinstalled to the latest AAH 2.2 which uses asterisk 1.2.1.
now I get recordings like this - these are all recorded with "record 
always" turn on in AMP.


internal to internal call - 20060109-212748-1136863668.139.WAV
incoming call - g200-20060109-205540-1136861730.135.WAV
outgoing call - OUT202-20060109-222003-1136866803.141.WAV
outgoing call - OUT203-20060109-205232-1136861552.133.WAV

here is the code I'm using now
--- snip ---
foreach($a as $b)
{
$k = explode(".", $b);
$l = explode("-", $k[0]);
$m = $k[1];
if (isset($l[3])) {
$unixtime = $l[3];
$o = $l[0];
} else {
$unixtime = $l[2];
$o = "internal";
}
$q = date('F j, Y \a\t g:i a', $unixtime);
echo "$i. $o made a call at $q, on channel 
".$m."  -

--- end snip ---

I'm guessing $m is a channel number or something like that since I don't 
have any extensions like 133, 135 or 139 -seems to be incrementing up

so it probably isn't a usefull number.

Your file names seem to have more usefull info - mine in AAH just have 
the date and time twice and the sip extension that made an outgoing 
call. There is probably a compile option somewhere that would set it 
like yours that I need to find.










My monitor filenames include the date and time, embedded as seconds 
since epoch iirc:


[EMAIL PROTECTED] monitor]$ ll
auto-1136394539-112-7476011-in.wav
auto-1136394539-112-7476011-out.wav
[EMAIL PROTECTED] monitor]$ ll
auto-1136394539-112-7476011.wav

So the 1136394539 part is seconds since epoch, 112 is who started the 
recording, 7476011 is where they were connected to when it happened.
And, I suspect the auto- part is 'cause I used automon feature to do 
this?  I haven't looked at asterisk code enough to see what filenames 
are created when.


Thank you for the patch though.  Now that I know many people are 
trying this stuff, I'll try to incorporate autodetection of filename 
style


Moj






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--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
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Re: [Asterisk-Users] VMauthenticate always asks for mailbox

2006-01-10 Thread C F
Can you post your voicemail.conf as well?

On 1/10/06, Gil Kloepfer <[EMAIL PROTECTED]> wrote:
> > On 1/10/06, Gil Kloepfer <[EMAIL PROTECTED]> wrote:
> > > The problem is that it always gives the "comedian mail" prompt and
> > > requests the mailbox number, even though I provide the mailbox
> > > number already.
>
> On Tue, Jan 10, 2006 at 10:00:44AM -0500, C F wrote:
> > Are you supplying the context?
>
> Actually I wasn't (according to the "help" it is optional).  However,
> even with the context it is behaving the same way.
>
> Is it correctly working for you?  If so, would you mind including an
> example of what you're doing so I can try that?
>
> Here's a trace on my PBX:
>
>-- Executing Goto("SIP/3771-9210", "Features|11945485|1") in new stack
>-- Goto (Features,11945485,1)
>-- Executing Macro("SIP/3771-9210", 
> "TestFeatureEnabled|5485|CallFwdUncond") in new stack
>-- Executing GotoIf("SIP/3771-9210", "1?:featnotenb") in new stack
>-- Executing GotoIf("SIP/3771-9210", "1?:featnotenb") in new stack
>-- Executing NoOp("SIP/3771-9210", "CallFwdUncondIsEnabled") in new stack
>-- Executing Answer("SIP/3771-9210", "") in new stack
>-- Executing VMAuthenticate("SIP/3771-9210", "[EMAIL PROTECTED]") in new 
> stack
>-- Playing 'vm-login' (language 'en')
> Jan 10 09:24:33 WARNING[8797]: app_voicemail.c:4912 vm_authenticate: Couldn't 
> read username
>  == Spawn extension (Features, 11945485, 3) exited non-zero on 'SIP/3771-9210'
>
> (I hung-up on it after it asked for the mailbox)
>
> Note that [EMAIL PROTECTED] exists and has a password.
>
> ---
> Gil Kloepfer
> [EMAIL PROTECTED]
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Re: [Asterisk-Users] CHAN_CAPI problem

2006-01-10 Thread Armin Schindler
I suggest you use the newer chan_capi-cm (loadable from sourceforge.net).

Armin

On Tue, 10 Jan 2006 [EMAIL PROTECTED] wrote:
> Hi all,
> I installed asterisk stable cvs 1.2 and chan_capi 0.4.0 PRE1, with one AVM
> Fritz Card ISDN connected to a Telecom NT1 Plus
> 
> I configured asterisk via AMP.
> No problem in making calls.
> If I try to ring the ISDN Phone Number, I don't see anything on the
> asterisk Console,
> I I activate the capi debug , I see the ring on the capi channel.
> If the context were wrong , I anyway should  see some line about this
> 
> Why I cannot see anything on asterisk ,nor in the /var/log/asterisk/full  ?
> 
> here is my /etc/asterisk/capi.conf
> 
> asteriskge03:/etc/asterisk # cat capi.conf
> ;
> ; CAPI config
> ;
> ;
> 
> ; general section
> 
> [general]
> nationalprefix=0
> internationalprefix=00
> rxgain=0.8
> txgain=0.8
> ;ulaw=yes;set this, if you live in u-law world instead of a-law
> 
> ; interface sections ...
> 
> [BRI1]  ;this example interface gets name 'ISDN1' and may be any
>  ;name not starting with 'g' or 'contr'.
> ;ntmode=yes  ;if isdn card operates in nt mode, set this to yes
> isdnmode=msn ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial)
>  ;when using NT-mode, 'DID' should be set in any case
> incomingmsn=*;allow incoming calls to this list of MSNs/DIDs, * = any
> ;defaultcid=123  ;set a default caller id to that interface for dial-out,
>  ;this caller id will be used when dial option 'd' is set.
> ;controller=0;ISDN4BSD default
> ;controller=7;ISDN4BSD USB default
> controller=1 ;capi controller number to use
> group=1  ;dialout group
> ;prefix=0;set a prefix to calling number on incoming calls
> softdtmf=on  ;enable/disable software dtmf detection, recommended for
> AVM cards
> relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf
> detection
> accountcode= ;Asterisk accountcode to use in CDRs
> context=from-pstn  ;context for incoming calls
> holdtype=hold;when Asterisk puts the call on hold, ISDN HOLD will be
> used. If
>  ;set to 'local' (default value), no hold is done and
> Asterisk may
>  ;play MOH.
> ;immediate=yes   ;DID: immediate start of pbx with extension 's' if no
> digits were
>  ; received on incoming call (no destination number
> yet)
>  ;MSN: start pbx on CONNECT_IND and don't wait for
> SETUP/SENDING-COMPLETE.
>  ; info like REDIRECTINGNUMBER may be lost, but this is
> necessary for
>  ; drivers/pbx/telco which does not send SETUP or
> SENDING-COMPLETE.
> ;echosquelch=1   ;_VERY_PRIMITIVE_ echo suppression
> ;echocancel=yes  ;EICON DIVA SERVER (CAPI) echo cancelation
>  ;(possible values: 'no', 'yes', 'force', 'g164', 'g165')
> echocancelold=yes;use facility selector 6 instead of correct 8 (necessary
> for older eicon drivers)
> ;echotail=64 ;echo cancel tail setting
> ;bridge=yes  ;native bridging (CAPI line interconnect) if available
> ;callgroup=1 ;Asterisk call group
> devices=2;number of concurrent calls on this controller
>  ;(2 makes sense for single BRI, 30 for PRI)
> 
> thanks in advance,
> Andrea
> 
> Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla.
> 
> Visitate il sito http://www.frameweb.it
> 
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[Asterisk-Users] does anyone know how to use 1.2 CVS setgroup in CAGI script

2006-01-10 Thread Raymond Chen








Dear all,

 

Any one has experience in CAGI script setgroup?  Please
let me know a bit of command detail.

 

Thanks,

 

Ray






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Re: [Asterisk-Users] 32 E1's in one Asterisk 'box'

2006-01-10 Thread Erick Perez
Well, this product from signate uses infiniband...
It has 4 slots for quad e1/t1 per slot.
http://www.signate.com/pdf/TelephonyServer.pdf
 
Just read the PDF. Obviously this is not an x86 Pc. I wonder if you want to build your own or were looking for a beast like this.
 
BTW, any real world comments about the signate machine? 
just a penny.
 
On 1/10/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]
> wrote:
The question was actually if Asterisk could support this and still actas one entity?I would in this case use 2 PC's with 16 E1's each, but this is as far as
I can see 2 separate PABX's. Are there a possibility to make these 2 (3or 4 or whatsoever) PC's act as one entity so I can connect a B channelon one PC to a B channel on another PC etc.JanAaron Daniel wrote:
> Wow, I agree with Alexander... putting that many lines in a single box> is one rather large single point of failure...>> Aaron>> Alexander Lopez wrote:>>>  I would look at using serveral machine splitting up the load using one
>> 4 port card in each.>>> -Original Message->>> From: [EMAIL PROTECTED]
>>> [mailto:[EMAIL PROTECTED]] On Behalf Of>>> [EMAIL PROTECTED]
>>> Sent: Tuesday, January 10, 2006 6:03 AM>>> To: Asterisk Users Mailing List - Non-Commercial Discussion>>> Subject: [Asterisk-Users] 32 E1's in one Asterisk 'box'>>>
>>> hi,>> My apologies for repeating this question, but I hoped re-frasing it>>> might help.>> I would like to assemble an PABX larger than what you possible can
>>> put inside one Asterisk box. What is the best way to do this? Can it>>> be done at all with Asterisk? Any ideas or hints would be apreaciated.>> jvb>>> ___
>>> --Bandwidth and Colocation provided by Easynews.com -->> Asterisk-Users mailing list>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users> ___
>> --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list>> To UNSUBSCRIBE or update options visit:>>
http://lists.digium.com/mailman/listinfo/asterisk-users>> ___> --Bandwidth and Colocation provided by 
Easynews.com -->> Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:>   http://lists.digium.com/mailman/listinfo/asterisk-users
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  http://lists.digium.com/mailman/listinfo/asterisk-users-- ---
Erick PerezLinux User 376588http://counter.li.org/  (Get counted!!!)Panama, Republic of Panama
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Re: [Asterisk-Users] Disconnected calls

2006-01-10 Thread Morten Isaksen

On 1/10/06, Kevin P. Fleming <[EMAIL PROTECTED]> wrote:
Morten Isaksen wrote:> We have some problems with calls that get disconnected in the middle of a
> call.>> We are using Asterisk 1.2.1 with a TE410P (2.gen firmware).>> When the call is disconnected Asterisk writes this to the log:> Jan 9 14:56:17 DEBUG[4404] dsp.c: ast_dsp_busydetect detected busy, avgtone:
> 300, avgsilence 2090Don't use 'callprogress=yes'.
 
 
callprogress=yes is commented out in zapata.conf so I dont think this is enabled.
 
busydetect was enabled. I have disabled this now, and is waiting for the users to report in.
 
-- Morten Isaksenhttp://www.misak.dk/blog/ 
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[Asterisk-Users] CHAN_CAPI problem

2006-01-10 Thread asterisk
Hi all,
I installed asterisk stable cvs 1.2 and chan_capi 0.4.0 PRE1, with one AVM
Fritz Card ISDN connected to a Telecom NT1 Plus

I configured asterisk via AMP.
No problem in making calls.
If I try to ring the ISDN Phone Number, I don't see anything on the
asterisk Console,
I I activate the capi debug , I see the ring on the capi channel.
If the context were wrong , I anyway should  see some line about this

Why I cannot see anything on asterisk ,nor in the /var/log/asterisk/full  ?

here is my /etc/asterisk/capi.conf

asteriskge03:/etc/asterisk # cat capi.conf
;
; CAPI config
;
;

; general section

[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
;ulaw=yes;set this, if you live in u-law world instead of a-law

; interface sections ...

[BRI1]  ;this example interface gets name 'ISDN1' and may be any
 ;name not starting with 'g' or 'contr'.
;ntmode=yes  ;if isdn card operates in nt mode, set this to yes
isdnmode=msn ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial)
 ;when using NT-mode, 'DID' should be set in any case
incomingmsn=*;allow incoming calls to this list of MSNs/DIDs, * = any
;defaultcid=123  ;set a default caller id to that interface for dial-out,
 ;this caller id will be used when dial option 'd' is set.
;controller=0;ISDN4BSD default
;controller=7;ISDN4BSD USB default
controller=1 ;capi controller number to use
group=1  ;dialout group
;prefix=0;set a prefix to calling number on incoming calls
softdtmf=on  ;enable/disable software dtmf detection, recommended for
AVM cards
relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf
detection
accountcode= ;Asterisk accountcode to use in CDRs
context=from-pstn  ;context for incoming calls
holdtype=hold;when Asterisk puts the call on hold, ISDN HOLD will be
used. If
 ;set to 'local' (default value), no hold is done and
Asterisk may
 ;play MOH.
;immediate=yes   ;DID: immediate start of pbx with extension 's' if no
digits were
 ; received on incoming call (no destination number
yet)
 ;MSN: start pbx on CONNECT_IND and don't wait for
SETUP/SENDING-COMPLETE.
 ; info like REDIRECTINGNUMBER may be lost, but this is
necessary for
 ; drivers/pbx/telco which does not send SETUP or
SENDING-COMPLETE.
;echosquelch=1   ;_VERY_PRIMITIVE_ echo suppression
;echocancel=yes  ;EICON DIVA SERVER (CAPI) echo cancelation
 ;(possible values: 'no', 'yes', 'force', 'g164', 'g165')
echocancelold=yes;use facility selector 6 instead of correct 8 (necessary
for older eicon drivers)
;echotail=64 ;echo cancel tail setting
;bridge=yes  ;native bridging (CAPI line interconnect) if available
;callgroup=1 ;Asterisk call group
devices=2;number of concurrent calls on this controller
 ;(2 makes sense for single BRI, 30 for PRI)

thanks in advance,
Andrea

Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla.

Visitate il sito http://www.frameweb.it

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[Asterisk-Users] Sacramento Asterisk Users Group

2006-01-10 Thread trixter aka Bret McDanel
http://www.sacaug.org/

At our new location 7-9pm Janurary 12th.  No fee to attend
Exit Certified 
8950 Cal Center Drive 
Suite 110, Bldg. 1 
Sacramento, California, USA 
95826

map links on the sacaug webpage.

We will be doing an install of astlinux with simple configuration as
well as discussing upcoming contests for which we have prizes donated by
Digium and TheVoipConnection.com it should be fun :)


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group


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[Asterisk-Users] Re: 32 e1's with asterisk

2006-01-10 Thread [EMAIL PROTECTED]


MvPhone wrote:


Hi,

You can put max of 2 4xE1 cards per dual CPU box. That means that you
end up with 4 dual CPU boxes

4 boxes * 2 cards * 4 E1 = 32 E1's

I sell affordable quadE1 cards that work well with asterisk. You can
check on pbxhardware.com

As for the dual CPU boxes it depends what will you do with the traffic
/ calls ... Are you planing to offer SIP termination with codec
transcoding ?

Will you have 32 physical E1 circuits ?
 


Yes!


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[Asterisk-Users] Another cisco question

2006-01-10 Thread Aaron Daniel
Sorry about the unrelated questions about cisco phones, but does anyone 
know how to set the second line up as a speed dial in the config file? 
Or is that specifically a per-user basis setting?


Aaron
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Re: [Asterisk-Users] MTU and Voice Delay (latency??)

2006-01-10 Thread Rusty Dekema
Oh, wow, ok. Doesn't look like the problem is with your WAN then! (Assuming that the ping times stay like that when the network is at its normal load.)  -RustyOn 1/10/06, 
Geoff Manning <[EMAIL PROTECTED]> wrote:
Rusty Dekema wrote:> How far (physically) is the Asterisk server location from the> location of the phones? Have you tried pinging the Asterisk server> from the network to which the phones are connected?
>> As a rule of thumb, If the two sites are within 2500 miles of each> other and the network connection between them is working properly,> the round trip time for a 64 byte ping should be less than 100 ms,
> the round trip time should not vary from one ping to another by more> than 2-5 ms (typical), and there should be virtually no dropped> packets (well under 0.1%).The two locations are in Greater London
Here is a traceroute from the Modem to the Asterisk Server (64byte packets)(snip)
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Re: [Asterisk-Users] Zaptel errors (power alarm?)

2006-01-10 Thread Moises Silva
Sorry Micheal, i have seen similar posts before, but no answers. May
be the best way to go is contact digium support.

Good Look

On 1/9/06, Michael Loftis <[EMAIL PROTECTED]> wrote:
> We've been having lost dialtone problems on one of our analog station
> ports.  Just before rebooting this time I noticed these in our dmesg
> outputonce the PBX comes back I'll get the times, but I can't help but
> think this must have something to do with it.  Anyone?  Do we need to have
> digium send us a replacement part?
>
> Ouch, part reset, quickly restoring reality (1)
> Power alarm on module 2, resetting!
> Ouch, part reset, quickly restoring reality (1)
> Power alarm on module 2, resetting!
> Ouch, part reset, quickly restoring reality (1)
> Power alarm on module 2, resetting!
> Ouch, part reset, quickly restoring reality (1)
> Power alarm on module 2, resetting!
> zaptel Disabled echo canceller because of tone (rx) on channel 23
> zaptel Disabled echo canceller because of tone (rx) on channel 23
> Ouch, part reset, quickly restoring reality (1)
> Power alarm on module 2, resetting!
> zaptel Disabled echo canceller because of tone (rx) on channel 23
>
>
> --
> "Genius might be described as a supreme capacity for getting its possessors
> into trouble of all kinds."
> -- Samuel Butler
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[Asterisk-Users] Re: using a Gigaset SX440isdn on a Diva 4BRI ?

2006-01-10 Thread Armin Schindler
On Tue, 10 Jan 2006, Louis-David Mitterrand wrote:
> On Mon, Jan 09, 2006 at 03:50:55PM +0100, Armin Schindler wrote:
> Here is the mlog with an attempt to Dial(CAPI/DIVA2/146472131):
> 
>   [C:4] 22:0167:202 - D-X(003) 02 01 7F
>   [C:4] 22:0168:202 - D-X(003) 02 01 7F
>   [C:4] 22:0169:202 - D-X(003) 02 01 7F
>   [C:4] 22:0170:202 - D-X(003) 02 01 7F
>   [C:4] 22:0171:201 - MDL-ERROR(G)
>   [C:4] 22:0171:202 - SIG-EVENT  0A
> 
>   [C:4] 22:0172:202 - D-X(003) 02 01 7F
>   [C:4] 22:0173:202 - D-X(003) 02 01 7F
>   [C:4] 22:0174:202 - D-X(003) 02 01 7F
>   [C:4] 22:0175:202 - D-X(003) 02 01 7F
>   [C:4] 22:0176:201 - MDL-ERROR(G)
>   [C:4] 22:0176:202 - SIG-EVENT  0A
> 
>   [C:4] 22:0177:202 - D-X(003) 02 01 7F
>   [C:4] 22:0178:202 - D-X(003) 02 01 7F
>   [C:4] 22:0179:202 - D-X(003) 02 01 7F
>   [C:4] 22:0180:202 - D-X(003) 02 01 7F
>   [C:4] 22:0181:201 - MDL-ERROR(G)
>   [C:4] 22:0181:202 - SIG-EVENT  0A
> 
>   [C:4] 22:0182:202 - D-X(003) 02 01 7F
>   [C:4] 22:0183:201 - D-X(003) 02 01 7F
>   [C:4] 22:0183:846 - CREATEID ok: context:1f assigned Id:5 freeIds=ec
>   [C:4] 22:0183:847 - alloc cr in use =3
>   [C:4] 22:0183:848 - [20] N-ASSIGN REQ Id:NL_ID, Ch:00
>   [C:4] 22:0183:848 - CREATEID ok: context:3f assigned Id:6 freeIds=eb
>   [C:4] 22:0183:848 - B2Assign  01 -- -- -- -- -- -- -- -- -- -- -- -- -- 
> -- --
>   [C:4] 22:0183:848 - B2Assign 1 Sig=05 d_id=01
>   [C:4] 22:0183:848 - Assign L2:2 L3:4 free:2389/2400 l:160 d:25480 
> fc:16000/15500
>   [C:4] 22:0183:848 - Layer 2 Transparent
>   [C:4] 22:0183:849 - [21] N-RC=ASSIGN_OK Id:06, Ch:00
>   [C:4] 22:0183:850 - [1,0] dsp_assign 0016, 0, 160
>   [C:4] 22:0183:850 - [1,0] CAI[00] 16 00 00 00 a0 00
>   [C:4] 22:0183:850 - [1,0] Download 512 requested
>   [C:4] 22:0183:851 - MORE
>   [C:4] 22:0183:851 - SIG-X(043) 08 01 11 05 A1 04 03 80 90 A3 18 01 81 
> 1E 02 80 83 6C 0C 00 80 30 31 34 36 34 37 32 31 33 30 70 0A 80 31 34 36 34 37 
> 32 31 33 31
>Q.931  CR11 SETUP
>   Sending complete
>   Bearer 
> Capability 80 90 a3
>   Channel Id 81
>   Progress 
> Indicator 80 83
>   Calling Party 
> Number 00 80 '0146472130'
>   Called Party 
> Number 80 '146472131'
>   [C:4] 22:0183:852 - SIG-S 0->1 e:885
>   [C:4] 22:0184:202 - D-X(003) 02 01 7F
>   [C:4] 22:0185:202 - D-X(003) 02 01 7F
>   [C:4] 22:0186:201 - MDL-ERROR(G)
>   [C:4] 22:0186:202 - SIG-EVENT  0A
> 
>   [C:4] 22:0186:202 - SIG-EVENT  0A
> 
>   [C:4] 22:0186:203 - EVENT: Call failed in State 'Call initiated'
>Link disconnected, TEI error
>   [C:4] 22:0186:203 - SIG-S 1->0 e:
>   [C:4] 22:0186:204 - [1,0] dsp_release
>   [C:4] 22:0186:207 - [22] N-REMOVE REQ Id:06, Ch:00
>   [C:4] 22:0186:208 - B2Release  01 05 -- -- -- -- -- -- -- -- -- -- -- 
> -- -- --
>   [C:4] 22:0186:208 - B2Release 1, free:2391, nomsg:0
>   [C:4] 22:0186:208 - Release - Layer 2 Transparent
>   [C:4] 22:0186:208 - DELETEID ok: deleted Id:6 freeIds=eb
>   [C:4] 22:0186:208 - [23] N-RC=OK Id:06, Ch:00
>   [C:4] 22:0186:209 - free cr in use =2
>   [C:4] 22:0186:209 - DELETEID ok: deleted Id:5 freeIds=ec
>   [C:4] 22:0187:202 - D-X(003) 02 01 7F
>   [C:4] 22:0188:202 - D-X(003) 02 01 7F
>   [C:4] 22:0189:202 - D-X(003) 02 01 7F
>   [C:4] 22:0190:202 - D-X(003) 02 01 7F
>   [C:4] 22:0191:201 - MDL-ERROR(G)
>   [C:4] 22:0191:202 - SIG-EVENT  0A
> 

The diva card is sending (D-X), but does not receive anything (D-R). It 
looks like either the cross connection still isn't working or the protocol
is wrong.

Armin
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Re: [Asterisk-Users] Problem with Chan_zap.so

2006-01-10 Thread Moises Silva
it seems to me that you need to upgrade your libpri, but i tought that
your chan_zap compilation should have died if you have not the correct
version of libpri. Anyway, what version do you have of libri? just
upgrade libpri and recompile chan_zap, or all asterisk just in case.

regards

On 1/9/06, Arinze Izukanne <[EMAIL PROTECTED]> wrote:
> I just upgraded to Asterisk 1.2.1 and Asterisk fails
> to start with the error below.
>
> Jan  9 21:25:38 NOTICE[1339]: cdr.c:1171 do_reload:
> CDR simple logging enabled.
> Jan  9 21:25:38 WARNING[1339]: loader.c:326
> __load_resource:
> /usr/lib/asterisk/modules/chan_zap.so: undefined
> symbol: pri_restart
> Jan  9 21:25:38 WARNING[1339]: loader.c:555
> load_modules: Loading module chan_zap.so failed!
>
>
> Can Anyone tell me whats wrong.
>
>
> A. Izukanne
>
> Atani Communications
>
>
>
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