date:20060519


On Tue, 16 May 2006, Tim Sharp [EMAIL PROTECTED] wrote

Chris,
When I tried background it waited until the message was done before 
dialing, just like playback.  Am I missing something?


Wasn't my suggestion :)

If I've understood what you're trying to I would go one of two ways:

Rather than dial each of the four numbers sequentially, dial them 
simultaneously. This should hopefully speed up your average pick up 
time, but will loose any control over preference for who deals with the 
call.


Or investigate queues. I don't have enough people to make it worth my 
while looking at these, so I've no idea if they're what you need, but 
they sound like they might be.

--
Chris Hastie
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[Asterisk-Users] Max Number of Extensions

2006-05-19 Thread Matteo Piazza


Hi,
I have a problem with the extensions file, asterisk load only 412 
extensions. I use 1.2.4. version of Asterisk, I have try to modify the 
extensions but asterisk finish to load the extensions when arrived at 
the 412th extension.


Matteo Piazza


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[Asterisk-Users] Experience with IBM X346 machines and Sangoma

2006-05-19 Thread Wolfgang Zweimueller


Hi All,

I have read many posts about problems with Asterisk on some systems. I
also set up Asterisk on many different boxes. But I have never seen
the following...

There is an IBM X346 (3.4GHz Xeon) with one Sangoma A104. This system
is currently idle, that means there is nothing running except Asterisk
(1.2.7.1). We are handling no calls now, but if I do a vmstat, I get
peaks in system load up to 40%! Here is an example:

procs ---memory-- ---swap-- -io --system-- cpu
 r  b   swpd   free   buff  cache   si   sobibo   incs us sy id wa
 0  0  0 375080 161660 14223200 0 0 4160   187  0  4 96  0
 0  0  0 375080 161660 14223200 0 0 4251   207  0  1 98  0
 0  0  0 375080 161660 14223200 0 0 4205   179  0  9 92  0
 0  0  0 375080 161660 14223200 036 4151   217  0  3 97  0
 0  0  0 375080 161660 14223200 0 0 4026   187  0  0 100  0
 0  0  0 375080 161660 14223200 0 0 4042   205  0 14 86  0
 0  0  0 375080 161660 14223200 0 0 4019   184  0 38 63  0
 0  0  0 375080 161660 14223200 0 0 4062   208  0  0 100  0
 0  0  0 375080 161660 14223200 0 0 4028   196  0  2 99  0
 0  0  0 375080 161660 14223200 016 4075   223  0 19 81  0
 1  0  0 375080 161660 14223200 0 0 4029   197  0  0 100  0
 0  0  0 375080 161660 14223200 0 0 4043   199  0  1 99  0
 0  0  0 375080 161660 14223200 0 0 4045   194  0  6 94  0
 0  0  0 375080 161660 14223200 0 0 4032   196  0 24 77  0
 0  0  0 375080 161660 14223200 012 4045   212  0  0 100  0
 0  0  0 375080 161660 14223200 0 0 4028   188  0  0 100  0



In contrast to the above I have a Dell 2850 running Asterisk and a lot
of other things (but no PRI card). This box is (according to vmstat)
almost always 100% idle!


Is anyone running a similar X346-system? What is the load and how does
Asterisk behave on it? Can anyone explain what is happening here?


Thx,
Wolfgang
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[Asterisk-Users] X100P not recognised on FreeBSD system

I've just received an OEM Wildcard X100P FXO card. Installing into my 
FreeBSD 5.4-RELEASE box it doesn't appear to be recognised at all. Since 
it's the first time I've put a PCI card in this machine I've just 
dropped a Netgear ethernet card in to make sure there isn't something 
fundamentally wrong with the motherboard, but that works fine.


Is there anything else I should check / try before assuming the X100P is 
faulty?


Output of pciconf -l -v below (after the Netgear card went back to where 
it belongs):


[EMAIL PROTECTED]:0:0:  class=0x06 card=0x31161106 chip=0x31161106 rev=0x00 
hdr=0x00
vendor   = 'VIA Technologies Inc'
device   = 'VT8375 ProSavageDDR PM266/KM266 CPU to PCI Bridge'
class= bridge
subclass = HOST-PCI
[EMAIL PROTECTED]:1:0: class=0x060400 card=0x0080 chip=0xb0911106 rev=0x00 
hdr=0x01
vendor   = 'VIA Technologies Inc'
device   = 'VT8633 Apollo Pro 266 CPU to AGP Controller'
class= bridge
subclass = PCI-PCI
[EMAIL PROTECTED]:16:0:class=0x0c0300 card=0x30381106 chip=0x30381106 
rev=0x80 hdr=0x00
vendor   = 'VIA Technologies Inc'
device   = 'VT82x UHCI USB 1.1 Controller (All VIA Chipsets)'
class= serial bus
subclass = USB
[EMAIL PROTECTED]:16:1:class=0x0c0300 card=0x30381106 chip=0x30381106 
rev=0x80 hdr=0x00
vendor   = 'VIA Technologies Inc'
device   = 'VT82x UHCI USB 1.1 Controller (All VIA Chipsets)'
class= serial bus
subclass = USB
[EMAIL PROTECTED]:16:2:class=0x0c0300 card=0x30381106 chip=0x30381106 
rev=0x80 hdr=0x00
vendor   = 'VIA Technologies Inc'
device   = 'VT82x UHCI USB 1.1 Controller (All VIA Chipsets)'
class= serial bus
subclass = USB
[EMAIL PROTECTED]:16:3:class=0x0c0320 card=0x73801462 chip=0x31041106 
rev=0x82 hdr=0x00
vendor   = 'VIA Technologies Inc'
device   = 'VT6202 USB 2.0 Enhanced Host Controller'
class= serial bus
subclass = USB
[EMAIL PROTECTED]:17:0:class=0x060100 card=0x31771106 chip=0x31771106 
rev=0x00 hdr=0x00
vendor   = 'VIA Technologies Inc'
device   = 'VT8235 PCI to ISA Bridge'
class= bridge
subclass = PCI-ISA
[EMAIL PROTECTED]:17:1:  class=0x01018a card=0x73801462 chip=0x05711106 
rev=0x06 hdr=0x00
vendor   = 'VIA Technologies Inc'
device   = 'VT82 EIDE Controller (All VIA Chipsets)'
class= mass storage
subclass = ATA
[EMAIL PROTECTED]:17:5:class=0x040100 card=0x73801462 chip=0x30591106 
rev=0x50 hdr=0x00
vendor   = 'VIA Technologies Inc'
device   = 'VT8233/33A/8235/8237 AC97 Enhanced Audio Controller'
class= multimedia
subclass = audio
[EMAIL PROTECTED]:18:0:  class=0x02 card=0x738c1462 chip=0x30651106 
rev=0x74 hdr=0x00
vendor   = 'VIA Technologies Inc'
device   = 'VT6102 Rhine II PCI Fast Ethernet Controller'
class= network
subclass = ethernet
[EMAIL PROTECTED]:0:0: class=0x03 card=0x73891462 chip=0x8d045333 rev=0x00 
hdr=0x00
vendor   = 'S3 Graphics Co., Ltd.'
device   = '86C420 ProSavage DDR'
class= display
subclass = VGA

--
Chris Hastie
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[Asterisk-Users] Call Transfer does not work

2006-05-19 Thread jbauer

Hi !

I am trying to transfer calls between internal SIP softclients, but it does
not work. Every time I press a key on the softclient, the CLI shows the
following output:

Attempting native bridge of SIP/456-9ee0 and SIP/173-f586

This is my extensions.conf:

[macro-voicemail]
exten = s,1,Dial(${ARG1},5,Ttr)
exten = s,2,Goto(status-${DIALSTATUS},1)
exten = status-BUSY,1,VoiceMail(b${MACRO_EXTEN})
exten = status-BUSY,2,Playback(vm-goodbye)
exten = status-BUSY,3,Hangup()
exten = status-NOANSWER,1,VoiceMail(u${MACRO_EXTEN})
exten = status-NOANSWER,2,Playback(vm-goodbye)
exten = status-NOANSWER,3,Hangup()

[internal]
exten = _ZXZ,1,Macro(voicemail,SIP/${EXTEN})

And this is the part of the features.conf I changed (just uncommented that
part)

[featuremap]
blindxfer = #1 ; Blind transfer
disconnect = *0; Disconnect
automon = *1   ; One Touch Record
atxfer = *2; Attended transfer

None of the shortcuts in [featuremap] works.

What am I doing wrong?

Regards, Jens
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[Asterisk-Users] Development news :: Smarter medialess calls!

2006-05-19 Thread Olle E Johansson


Friends,
To update you on recent changes in svn trunk, I can inform you that  
we now have ever smarter
ways to handle media streams in Asterisk than we do in 1.2 for the  
IAX2 and SIP protocols.


* IAX2: Splitting signalling and media apart

Starting with the IAX2 protocol, we now have the ability to transfer  
media streams to go directly
between IAX2 servers and keep the signalling path. Before, when  
Asterisk did a native transfer
to optimize the IAX2 call path, we lost all tracks of the call and  
could not get a CDR. With this
patch, by Mark, we now have a hybrid solution that releases the media  
but keeps IAX2 signalling.
This is a very new feature, so I don't expect the various non- 
asterisk IAX2 clients out there to
support it yet. When they do, it will mean a huge change in the  
number of calls your server can

handle. For now, this optimizes calls in Asterisk IAX2 clusters.

* SIP: Removing the media immediately, not as an afterthought

Mark and Kevin have been working on various ways to optimize the  
setup of a SIP call
where Asterisk has no reason to stay in the media stream. Asterisk  
will now setup the
call directly between the two devices instead of accepting the call,  
staying in the stream and
then, as a sudden afterthought, send re-invites to release the media  
stream.


An additional new feature, inspired by a community patch on the bug  
tracker, is that
we now also release calls if SIP INFO dtmf is used. Since the DTMF is  
not handled in
the RTP media stream, we can release the call (unless there is  
another reason to stay

in the media path, like NAT support).

These changes optimize your Asterisk a great deal and will hopefully  
make Asterisk
scale a bit more. Your development team is as always focused on  
scaling issues, trying
to go where no software PBX has gone before, explore new telephony  
territories...

VoiP trekking... Well, enough of that. Sorry, got sidetracked.

* Asterisk 1.4 - I see a shape, an outline

The work with Asterisk 1.4 is going into the final stages. We are  
working hard to commit
the changes that are ready and finalize the 1.4 release. If you visit  
the bug tracker, you already
see patches that we've marked post 1.4 since we feel they're not  
ready. The next release is
not that far away, so it's not a big thing. We won't wait over 1 year  
like we did between 1.0 and

1.2.

This weekend, I'm leaving for my Training in New York. Next training  
is in Stockholm,
Sweden in June, after that we're launching the Asterisk SIP  
Masterclass in Chicago in

July - with a gold team teaching: Ed Guy, Terry Wilson and myself.

While I'm travelling around, you can spend all your free time testing  
Asterisk 1.4 for us.

We need your help, now. Download svn trunk and test in your environment!

On behalf of the community - thank you for testing!

SIP greetings!
/Olle

---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/



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[Asterisk-Users] Watchguard Firebox 1000 woes

2006-05-19 Thread Kerry Garrison




We are trying to 
setup a sip connection behind a Watchguard Firebox 1000 and it is simply 
put...not working. The ports are all forwarded but the packets are not going 
out. It is as if the firewall simply ignores SIP packets. Has anyone seen this 
or have any idea what the issue could be? Watchguard so far has been of zero 
help.

Kerry GarrisonDirector of 
Technical ServicesTech Data Pros - Orange County's Mobile IT Service 
Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com 


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RE: [Asterisk-Users] router with qos and compatible with stun

2006-05-19 Thread HaoXu

 
Hi Laurent,

From your message, I am not very clear what device do you want? Just a
ethernet router? For 20 simultaneous voip connections, it seems to use a FXO
gateway.

Hawk.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Laurent
Schweizer
Sent: Friday, May 19, 2006 3:55 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] router with qos and compatible with stun


Hello, 

I have a client that needs 20-26 simultaneous voip connections and I don't
want to relay all this traffic. So I m looking for a router with non
symmetric NAT for SDSL. (to use STUN)


Thanks


Laurent


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RE: [Asterisk-Users] monitoring sangoma cards via snmp

2006-05-19 Thread HaoXu

 

As I known, there are many gateway provide SNMP support.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sangoma
Techdesk
Sent: Friday, May 19, 2006 4:09 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] monitoring sangoma cards via snmp

When used in TDM mode, the sangoma cards will work under zaptel, so you will
need to perform SNMP at a higher level (i.e in Asterisk). 

David Yat Sin
Sangoma Technologies
(905) 474 1990 x119
(800) 388 2475 x199
MSN: [EMAIL PROTECTED]
Email: [EMAIL PROTECTED]
Wiki: http://sangoma.editme.com
 

 Message: 2
 Date: Fri, 12 May 2006 09:39:55 +0200 (CEST)
 From: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] monitoring sangoma cards via snmp
 To: asterisk-users@lists.digium.com
 Message-ID: [EMAIL PROTECTED]
 Content-Type: text/plain; charset=iso-8859-1
 
 Hello,
 
 Digium does not provide snmp support to monitor their cards !
 
 Anybody has tried Sangoma product A104 Quad T1/E1 or others ?
 
 Regards
 harry
 
 
 
 
 


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RE: [Asterisk-Users] Home asterisk system with single PSTN Line

2006-05-19 Thread HaoXu

 Right, a adapter will work fine than a card.  It can work independently and
also have some nice feature. Such as PAP2 could support FXS to FXO lifeline
when powe lose. The MG3000-R support auto reroute when voip down.

Hawk

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Forrest Beck
Sent: Friday, May 19, 2006 5:27 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Home asterisk system with single PSTN Line

All you would need is a analog adapter.  Grandstream makes a nice one that
acts as FXS (subscriber port) and FXO (office port).  Have a look at:

http://www.voipsupply.com/product_info.php?manufacturers_id=12products_id=4
51osCsid=8fa9170b8ddbef50ad06d859b85ba396

and

http://en.wikipedia.org/wiki/Fxo
http://en.wikipedia.org/wiki/Fxs


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Barrass Kevin
Sent: Thursday, May 18, 2006 9:53 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Home asterisk system with single PSTN Line


Hi

Im new to asterisk and want to setup a small system at home to play with.

Can anyone advise a good card I can use so the asterisk box Im building can
act as a gateway to PSTN using my single home analogue phone line.

Kind Regards

Kev
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Re: [Asterisk-Users] Max Number of Extensions

2006-05-19 Thread Alejandro Vargas


2006/5/19, Matteo Piazza [EMAIL PROTECTED]:

Hi,
I have a problem with the extensions file, asterisk load only 412
extensions. I use 1.2.4. version of Asterisk, I have try to modify the
extensions but asterisk finish to load the extensions when arrived at
the 412th extension.


Only some ideas: Check the file size and see to see if there is a
problem with it. Try removing the part corresponding to the last
extension that worked and the next to see if there is an error in the
file. ¿mixed lines created with windows and linux? ¿LF / CR-LF
problems? Try moving the last part to another file and include it.


--
Alejandro Vargas
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Re: [Asterisk-Users] SIP Header Info

2006-05-19 Thread Olle E Johansson



19 maj 2006 kl. 00.14 skrev Douglas Garstang:

Cool. Thanks. Now, I'm just wondering what SIP methods that will  
work on? Need to peek into a REFER message from a phone.


I don't think that will work, but it is a cool idea since REFER  
creates a new call and goes through the dial plan again.


/O
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[Asterisk-Users] help about modem

2006-05-19 Thread Muhammad atif amin

Hi,
 Sorry if question is stupid..As i m just new to asterisk..
I need help in the following schenerio..
Actually i want to transfer incoming call from PSTN to any PC in the LAN. 
Can i use modem for this purpose and also need help in configuration
for this schenerio.woul any one plz give configuration sample
reagarding my problem..

Thanks in advance

Best Regards
___
Atif Amin

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Re: [Asterisk-Users] Home asterisk system with single PSTN Line

2006-05-19 Thread Gareth Blades

I use a X100P from www.x100p.com and it works well. It is purely a FXO
card for connecting to a PSTN line though. I have a SIP phone aswell
which is why I dont require a FXS interface.

I have an account with voiptalk.org so I can make calls over the
internet. I also have a free local rate number with voipuser for
incoming calls.
We use Asterisk at work aswell so I have the two systems linked so when
working from home my home phone rings when someone phones my work
extension. My work phone rings when my home phone rings also :)


On Thu, 2006-05-18 at 14:52, Barrass Kevin wrote:
 Hi
 
 Im new to asterisk and want to setup a small system at home to play
 with.
 
 Can anyone advise a good card I can use so the asterisk box Im building
 can act as a gateway to PSTN using my single home analogue phone line.
 
 Kind Regards
 
 Kev
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[Asterisk-Users] Faxing with Asterisk using both ISDN and FXS

2006-05-19 Thread Benjamin SEBBAH

Hi everyone,I have a question about faxing. I am running
Asterisk SVN-trunk-r7498 on a ubuntu server and everything is going
fine. I have a tdm40b, a tdm04b and an avm fritz!card pci plugged on it.Here is what I'd like to do:
Receiving fax via the ISDN avm fritz!card, sending the fax via
email (for the moment I know I can do that) but also routing the fax
trough my old fax machine which would be plugged in an FXS port (which
means using the old fax machine as a printer).
Using the same old fax machine to send fax which means translating
the fax so that the ISDN card understand it and send it (which means
using the old fax machine as a scanner).Any idea if it is possible?
Thanks in advance,Benjamin SEBBAHAduneo France
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Re: [Asterisk-Users] Max Number of Extensions

2006-05-19 Thread Chen Fan

not heard that there have some extensions limitions.
On 5/19/06, Alejandro Vargas [EMAIL PROTECTED] wrote:
2006/5/19, Matteo Piazza [EMAIL PROTECTED]:
 Hi, I have a problem with the extensions file, asterisk load only 412 extensions. I use 1.2.4. version of Asterisk, I have try to modify the extensions but asterisk finish to load the extensions when arrived at
 the 412th extension.Only some ideas: Check the file size and see to see if there is aproblem with it. Try removing the part corresponding to the lastextension that worked and the next to see if there is an error in the
file. ¿mixed lines created with windows and linux? ¿LF / CR-LFproblems? Try moving the last part to another file and include it.--Alejandro Vargas___
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http://lists.digium.com/mailman/listinfo/asterisk-users-- Jeffery　　`∧ ∧︵ 　　ミ^r^ミ灬)~iaxtel Num: 1-700-576-1311fwdnet Num: 728150
http://www.diaip.com 
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[Asterisk-Users] voicemail access on the Thomson ST2030 ?

2006-05-19 Thread Louis-David Mitterrand

Hello,

After reading all the docs and going through the menus, I still can't 
find the voicemail access button or menu sequence on the ST2030 
(http://www.voip-info.org/wiki/view/Thomson+ST2030)

Also I can't get phone provisionning through tftp to work. Configuration 
files are loaded but the phone seems to ignore them.

Any idea?
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Re: [Asterisk-Users] X100P not recognised on FreeBSD system

2006-05-19 Thread Dinesh Nair




On 05/19/06 16:30 Chris Hastie said the following:
I've just received an OEM Wildcard X100P FXO card. Installing into my 
FreeBSD 5.4-RELEASE box it doesn't appear to be recognised at all. Since 


have you downloaded, compiled and installed the zaptel-bsd drivers ? if you 
haven't, instructions for getting them are at 
http://www.voip-info.org/wiki-FreeBSD+zaptel


for the X100P, you'd need to kldload zaptel.ko and wcfxo.ko, though be 
warned that the wcfxo.ko driver has not had much development in yonks.


--
Regards,   /\_/\   All dogs go to heaven.
[EMAIL PROTECTED](0 0)   http://www.openmalaysiablog.com/
+==oOO--(_)--OOo==+
| for a in past present future; do|
|   for b in clients employers associates relatives neighbours pets; do   |
|   echo The opinions here in no way reflect the opinions of my $a $b.  |
| done; done  |
+=+
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[Asterisk-Users] AsteriskOUT

2006-05-19 Thread [EMAIL PROTECTED]


Hi all,

Has anyone in the group tried the services of www.asteriskout.com. (lunaphone)

Just thought of letting you all be aware not to fall on their services
as it always seen attractive but to my experience they had always been
pulling the the legs of customers with an approach of making money
through ways that are thought to be non industrial.

Just wanted to warn you guys

Dan
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Re: [Asterisk-Users] X100P not recognised on FreeBSD system


On Fri, 19 May 2006, Dinesh Nair [EMAIL PROTECTED] wrote:




On 05/19/06 16:30 Chris Hastie said the following:
I've just received an OEM Wildcard X100P FXO card. Installing into 
my FreeBSD 5.4-RELEASE box it doesn't appear to be recognised at 
all. Since


have you downloaded, compiled and installed the zaptel-bsd drivers ? if you
haven't, instructions for getting them are at
http://www.voip-info.org/wiki-FreeBSD+zaptel

for the X100P, you'd need to kldload zaptel.ko and wcfxo.ko, though be
warned that the wcfxo.ko driver has not had much development in yonks.


Yes, I have these. The modules load, but ztcfg complains
ZT_CHANCONFIG failed on channel 1: No such device or address (6) and as I
said, it doesn't appear that the card has been recognised by the kernel.


--
Chris Hastie
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Re: [Asterisk-Users] Plan to free myself from AAH

2006-05-19 Thread Nils-Anders Duesund Nøttseter

Alex Epshteyn wrote:
 Please take a look at PBX Manager - you may find it flexible and easy to
 extend.
 
 http://www.thirdlane.com
Hi Alex. I see you're promoting your product :)

I've tried the online demo and it looks nice, but is there any
possibility of a demo version that I can download and test in my lab?

The reason I'd like that is to compare the output of PBX Manager to the
output of FreePBX. I need to be able to add custom configuration, and I
don't like the FreePBX way.

-- 
Nils-Anders Nøttseter
Linpro AS
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Re: [Asterisk-Users] X100P not recognised on FreeBSD system

2006-05-19 Thread Dinesh Nair



On 05/19/06 18:57 Chris Hastie said the following:

Yes, I have these. The modules load, but ztcfg complains
ZT_CHANCONFIG failed on channel 1: No such device or address (6) and as I
said, it doesn't appear that the card has been recognised by the kernel.


could you try the X100P in anther system to rule out issues with the Via 
board you're using ?


--
Regards,   /\_/\   All dogs go to heaven.
[EMAIL PROTECTED](0 0)   http://www.openmalaysiablog.com/
+==oOO--(_)--OOo==+
| for a in past present future; do|
|   for b in clients employers associates relatives neighbours pets; do   |
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Re: [Asterisk-Users] asterisk and ODBC

2006-05-19 Thread Dumpolid Exeplish

Hi,I have duetifully followed the instructions in the cdr.txt but asterisk still cannot connect to the MS SQL server. I can connect using the ODBC native connector bu asterisk still cannot
On 5/16/06, Bruce Reeves [EMAIL PROTECTED] wrote:
Yes you can use MSSQL with the ODBC driver, I have it working for CDR logs, I had to install unixODBC and configure it then use the cdr_odbc.conf file to specify. Check in your source files or svn checkout for a docs folder and the 
cdt.txt file.On 5/16/06, Dumpolid Exeplish 
[EMAIL PROTECTED] wrote:

Hi All,How can i use Microsoft SQL server with asterisk, Can the unix ODBC diriver interface MSQL?? and what module would i be using to access ODBC from asterisk??

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

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[Asterisk-Users] call recording - contrlo of Ast in 'h' extension

2006-05-19 Thread Lukasz Galaska

Hi,

I'm new on this list so on the beginning I'd like to
say 'hello' to everyone.

During my work with Asterisk(1.0.10) I've encountered
two problems. I was looking for solution on the web
but with no result:
- I work on optional call recording. Right now call is
recorded from the beginning. I wonder how I can
configure the dialplan that after the Caller hangs up
I can decide  to archive this call (don't remove the
voice file) or not (remove this file from the hard
drive). I tried multiple configurations but with no
success (the problem is with getting the control over
Asterisk after Caller hangs up).
- Another problem concerns recording of the call in
gsm or WAV49 format. In my dialplan recorded voice
file after hang up should be sent to ftp server. The
problem is that it takes a while Asterisk to do it.
Simply, the dialplan runs so fast that compression (I
guess) doesn't keep up with it. As a result only a
piece file (4KB) is uploaded to ftp.
By the way, when I switch to 'wav' format everything
goes smoothly and there's no problem.
I've tried to use Wait but it doesn't work with 'h'
extension (everything is happening after hang up).

I'd appreciate if anyone could help. I've spent lots
of time on it and run out of any ideas.

Luke 


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Re: [Asterisk-Users] Development news :: Smarter medialess calls!

2006-05-19 Thread Sean Cook

Olle,

Is there a poster of you that I can put up on my wall ;)

Regards,

Sean

Olle E Johansson wrote:
 Friends,
 To update you on recent changes in svn trunk, I can inform you that we
 now have ever smarter
 ways to handle media streams in Asterisk than we do in 1.2 for the
 IAX2 and SIP protocols.

 * IAX2: Splitting signalling and media apart

 Starting with the IAX2 protocol, we now have the ability to transfer
 media streams to go directly
 between IAX2 servers and keep the signalling path. Before, when
 Asterisk did a native transfer
 to optimize the IAX2 call path, we lost all tracks of the call and
 could not get a CDR. With this
 patch, by Mark, we now have a hybrid solution that releases the media
 but keeps IAX2 signalling.
 This is a very new feature, so I don't expect the various non-asterisk
 IAX2 clients out there to
 support it yet. When they do, it will mean a huge change in the number
 of calls your server can
 handle. For now, this optimizes calls in Asterisk IAX2 clusters.

 * SIP: Removing the media immediately, not as an afterthought

 Mark and Kevin have been working on various ways to optimize the setup
 of a SIP call
 where Asterisk has no reason to stay in the media stream. Asterisk
 will now setup the
 call directly between the two devices instead of accepting the call,
 staying in the stream and
 then, as a sudden afterthought, send re-invites to release the media
 stream.

 An additional new feature, inspired by a community patch on the bug
 tracker, is that
 we now also release calls if SIP INFO dtmf is used. Since the DTMF is
 not handled in
 the RTP media stream, we can release the call (unless there is another
 reason to stay
 in the media path, like NAT support).

 These changes optimize your Asterisk a great deal and will hopefully
 make Asterisk
 scale a bit more. Your development team is as always focused on
 scaling issues, trying
 to go where no software PBX has gone before, explore new telephony
 territories...
 VoiP trekking... Well, enough of that. Sorry, got sidetracked.

 * Asterisk 1.4 - I see a shape, an outline

 The work with Asterisk 1.4 is going into the final stages. We are
 working hard to commit
 the changes that are ready and finalize the 1.4 release. If you visit
 the bug tracker, you already
 see patches that we've marked post 1.4 since we feel they're not
 ready. The next release is
 not that far away, so it's not a big thing. We won't wait over 1 year
 like we did between 1.0 and
 1.2.

 This weekend, I'm leaving for my Training in New York. Next training
 is in Stockholm,
 Sweden in June, after that we're launching the Asterisk SIP
 Masterclass in Chicago in
 July - with a gold team teaching: Ed Guy, Terry Wilson and myself.

 While I'm travelling around, you can spend all your free time testing
 Asterisk 1.4 for us.
 We need your help, now. Download svn trunk and test in your environment!

 On behalf of the community - thank you for testing!

 SIP greetings!
 /Olle

 ---
 * Olle E. Johansson - [EMAIL PROTECTED]
 * Asterisk Training http://edvina.net/training/



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Re: [Asterisk-Users] asterisk and ODBC

2006-05-19 Thread Sean Cook

have you tested to make sure that you can connect to the odbc resource
outside of asterisk via perl/php/(insert random language here)?  Make
sure odbc is setup correctly and working before proceeding with the
asterisk part.


Sean

Dumpolid Exeplish wrote:
 Hi,
 I have duetifully followed the instructions in the cdr.txt but
 asterisk still cannot connect to the MS SQL server. I can connect
 using the ODBC native connector bu asterisk still cannot

 On 5/16/06, *Bruce Reeves* [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] wrote:

 Yes you can use MSSQL with the ODBC driver, I have it working for
 CDR logs, I had to install unixODBC and configure it then use the
 cdr_odbc.conf file to specify. Check in your source files or svn
 checkout for a docs folder and the cdt.txt file.


 On 5/16/06, *Dumpolid Exeplish*  [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] wrote:
 Hi All,
 How can i use Microsoft SQL server with asterisk, Can the unix
 ODBC diriver interface MSQL?? and what module would i be using to
 access ODBC from asterisk??


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 -- 
 Bruce
 Nortex Networks

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Re: [Asterisk-Users] Call detail records for Digital Receptionist

2006-05-19 Thread Patrick

On Fri, 2006-05-19 at 13:27 +0100, Robbie Hughes wrote:
 I have a client who uses the digital receptionist to take calls and then
 pass them on as normal to the appropriate depts. The problem is, they need
 to find out how many people are not getting through...
 In the mysql cdr, these are all recorded as ANSWERED, but there is no
 indication of if they give up on waiting.

I guess you could get the info by using queues and extract the
statistics how many people hung up.

Regards,
Patrick
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[Asterisk-Users] Snom firmwares suck

2006-05-19 Thread Remco Barende

Most people seem quite positive about Snom phones, I cannot share this 
opinion.


The displays are dying quite often, and firmware is buggy. I have tried 
every firmware from 4.5 up to 5.x and 6.04 but keep having problems with 
phones locking up or rebooting during an ongoing conversation.


REALLY annoying for a phone that is advertised / targeted as a business 
class phone

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[Asterisk-Users] SIP register problem

2006-05-19 Thread asterisk


Hi all,

We have two asterisk PBX. We would like to register it with SIP peer.
The client sends the register request. It gets back:
Jan  2 01:31:27 WARNING[186]: chan_sip.c:9760 handle_response_register: 
Got 404 Not found on SIP register to service [EMAIL PROTECTED], 
giving up


server:
Sip.conf
[general]
context=blackbox-in ; Default context for incoming calls
realm=xxx
bindport=5061   ; UDP Port to bind to (SIP standard port 
is 5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds 
to all)

srvlookup=yes   ; Enable DNS SRV lookups on outbound calls

[sipteszt]
type=user
context=siporka
username=sipteszt1
fromuser=sipteszt1
secret=password
accountcode=sipteszt
host=dynamic
disallow=all
;allow=ilbc
;allow=speex
allow=gsm
allow=speex
allow=alaw
nat=yes
notransfer=yes
canreinvite=no
qualify=no

[sipteszt]
type=peer
fromuser=sipteszt1
username=sipteszt1
secret=password
accountcode=sipteszt
host=dynamic
disallow=all
;allow=ilbc
;allow=speex
allow=gsm
allow=speex
allow=alaw
notransfer=yes
canreinvite=no
qualify=yes

Client:
sip.conf
[general]
context=default ; Default context for incoming calls
accountcode=sip
bindport=5060   ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes   ; Enable DNS SRV lookups on outbound calls
callevents=yes  ; generate manager events when sip ua performs 
events (e.g. hold)

subscribecontext=ext-local-custom

register = sipteszt:[EMAIL PROTECTED]/sipteszt

[sipteszt]
type=peer
host=217.xxx.32.207
fromuser=sipteszt1
fromdomain=
username=sipteszt1
secret=password
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=alaw
allow=gsm
allow=ilbc
nat=no
qualify=no
accountcode=12
trunktimestamps=no

; incoming peer from 217.xxx.32.207
[sipteszt-in]
type=user
host=217.xxx.32.207
username=sipteszt1
context=incoming
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=gsm
allow=ilbc
accountcode=12
trunktimestamps=no


Kind regards Szolke
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Re: [Asterisk-Users] AsteriskOUT

2006-05-19 Thread Ronald Wiplinger


[EMAIL PROTECTED] wrote:

Hi all,

Has anyone in the group tried the services of www.asteriskout.com. 
(lunaphone)


Just thought of letting you all be aware not to fall on their services
as it always seen attractive but to my experience they had always been
pulling the the legs of customers with an approach of making money
through ways that are thought to be non industrial.

Are you talking about BUSINESS 
If so, there is another mailing list available!
If it is a USERS question, and allow me please to quote the name of 
this email list:

   Asterisk Users Mailing List - Non-Commercial Discussion
than you may specify your problem so that we can try to help you.





Just wanted to warn you guys


wow!


bye

Ronald Wiplinger

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[Asterisk-Users] Forwarded Calls crash the system on 64 bit

2006-05-19 Thread Jonathan k. Creasy

I have a strange problem. I have a central server with my PRI on it.
There are three peripheral servers connected via IAX. 

I have a 64bit system for my central server and the backup system is a
32bit system. If I have forwarding (sip redirect) turned on and
forwarding to an outside number (i.e. my cell phone) when the 64bit
system is in the middle it will crash. The Asterisk process doesn't
actually crash so there is no backtrace. It uses about 99% of the CPU
and all IAX channels go down and IAX will no longer accept connections.
SIP calls continue as if nothing had happened although audio quality is
compromised due to the CPU being used heavily. A quick restart now
fixes it right up. If I bring the 32 bit system up and have it doing the
routing then there will be one-way audio on the forwarded call
(termination point can here the originator but the called cannot be
heard). It does not crash and all the other calls are unaffected. 

Both systems are CentOS 4.2 fully updated and running Asterisk 1.2.7.
The peripheral system is Fedora Core 3 running Asterisk 1.2.7 also. 

-Jonathan
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Re: [Asterisk-Users] Snom firmwares suck

2006-05-19 Thread Alex Robar

Interesting, I've never had a screen just die on me. Are you saying that the screen just stops working?As for firmware, I've always found that the best way to deal with problematic firmware is to talk to the company about it. Especially in a scenario like you're in, where everyone else seems happy with the firmware. It's very likely that Snom may not be aware of the problems you are experiencing. I can't say that I've ever had a Snom reboot in the middle of a conversation. I don't think my phones have ever just rebooted when I haven't asked them to. I would certainly talk to Snom about this.
AlexOn 5/19/06, Remco Barende [EMAIL PROTECTED] wrote:
Most people seem quite positive about Snom phones, I cannot share thisopinion.The displays are dying quite often, and firmware is buggy. I have triedevery firmware from 4.5 up to 5.x and 6.04 but keep having problems with
phones locking up or rebooting during an ongoing conversation.REALLY annoying for a phone that is advertised / targeted as a businessclass phone___--Bandwidth and Colocation provided by 
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Re: [Asterisk-Users] Snom firmwares suck

2006-05-19 Thread Steve Davies


On 5/19/06, Remco Barende [EMAIL PROTECTED] wrote:

Most people seem quite positive about Snom phones, I cannot share this
opinion.

The displays are dying quite often, and firmware is buggy. I have tried
every firmware from 4.5 up to 5.x and 6.04 but keep having problems with
phones locking up or rebooting during an ongoing conversation.

REALLY annoying for a phone that is advertised / targeted as a business
class phone


Hmmm... A random statement out of the blue... I assume that you meant
to add Does any kind soul have a suggestion to help out? :-)

I find that the snom phones can be over-sensetive to network glitches,
which with the default configuration can cause a reboot (usually
caused by cheap switches). Try changing the reboot on ethernet unplug
setting to ignore.

It helps here.
Steve
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Re: [Asterisk-Users] Snom firmwares suck

2006-05-19 Thread Dr. Michael J. Chudobiak


Remco Barende wrote:
Most people seem quite positive about Snom phones, I cannot share this 
opinion.


The displays are dying quite often, and firmware is buggy. I have tried 
every firmware from 4.5 up to 5.x and 6.04 but keep having problems with 
phones locking up or rebooting during an ongoing conversation.


REALLY annoying for a phone that is advertised / targeted as a business 
class phone


Remco,

I have a dozen Snom 360s. One had a defective LCD that would become 
garbled after time. Snom support quickly confirmed that this was a known 
issue, and my vendor (voipsupply) quickly sent a replacement.


I've never seen any lockups or reboots. I reboot the phones each night 
at midnight, just to be safe - try doing that to see if it reduces 
problems. I've posted a sample perl cron script at 
http://www.voip-info.org/wiki/view/Asterisk+phone+snom to show how.


I use shielded ethernet cables (STP) everywhere too. Try that - good 
grounding may be beneficial. It can't hurt, anyway.


Snom support is pretty responsive. Try emailing [EMAIL PROTECTED]; they 
have fixed some issues for me (for example, the clock was showing the 
wrong time due to daylight savings time problems).


Try using a Grandstream GXP-2000 phone, and you'll see why people like 
the Snoms :-)


Hope this helps - let us know if anything makes a difference!


- Mike
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RE: [Asterisk-Users] SIP useragent?

2006-05-19 Thread William Piper

If this is using realtime SIP database, you will need to put 

rtcachefriends = yes

into your sip.conf global settings to have it show with sip show peers


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remco Barende
Sent: Friday, May 19, 2006 3:38 AM
To: Asterisk Users List
Subject: [Asterisk-Users] SIP useragent?

Hi list !

Is it possible to show the used Useragent of a peer that 
registered with Asterisk? It's being saved obviously because the 
console says so when a phone is registering but sip show peers doesn't 
show it?

Is there any other way to view it?

Thanks!
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Re: [Asterisk-Users] Experience with IBM X346 machines and Sangoma

2006-05-19 Thread Evan Borgström


No Sagnoma hardware here but all of my machines are either IBM x345's
(2x2.8GHz / 4GB RAM) or x346's (2x3.0GHz / 4GB RAM). Here's a vmstat
from over a 2 minute period from a gateway machine that had 6 active
calls (but Asterisk was not transcoding and not in the media path).

# vmstat 1 1 | grep -v '100  0$'
procs ---memory-- ---swap-- -io --system--
cpu
 r  b   swpd   free   buff  cache   si   sobibo   incs us sy
id wa
 1  0164 233584  8 54817200 0 13 3  1  0
99  0
procs ---memory-- ---swap-- -io --system--
cpu
 r  b   swpd   free   buff  cache   si   sobibo   incs us sy
id wa
procs ---memory-- ---swap-- -io --system--
cpu
 r  b   swpd   free   buff  cache   si   sobibo   incs us sy
id wa
procs ---memory-- ---swap-- -io --system--
cpu
 r  b   swpd   free   buff  cache   si   sobibo   incs us sy
id wa
procs ---memory-- ---swap-- -io --system--
cpu
 r  b   swpd   free   buff  cache   si   sobibo   incs us sy
id wa
procs ---memory-- ---swap-- -io --system--
cpu
 r  b   swpd   free   buff  cache   si   sobibo   incs us sy
id wa
procs ---memory-- ---swap-- -io --system--
cpu
 r  b   swpd   free   buff  cache   si   sobibo   incs us sy
id wa
procs ---memory-- ---swap-- -io --system--
cpu
 r  b   swpd   free   buff  cache   si   sobibo   incs us sy
id wa


The only time it wasn't 100% idle is right when vmstat started.

-Evan

Cosmin Prund wrote:
 I get the same thing with a basic AAH (2.8 or whatever the latest is at
 the time of this writing). I consider the thing an pretty basic system
 running nothing but Asterisk yet I get the same kind of spikes once
 every 10 seconds (running vmstat 1 1). My system is a cheap Sepron
 2800+ on a ASUS MB with nVidia3 chipset. I'd love to know what's that
 all about...
 
 My sample below. As you can see I mostly have 100% idle and there are
 spikes :-)
 
 procs ---memory-- ---swap-- -io --system--
 cpu
 0  0160  60908  17888 36945200 0 0 2110   199  1  0
 99  0
 0  0160  60908  17888 36945200 0 0 2110   206  0  0
 100  0
 0  0160  60908  17896 36945200 012 2112   206  0  0
 100  0
 0  0160  60908  17896 36945200 0 0 2112   205  0 21
 79  0
 0  0160  60908  17896 36945200 0 0 2111   209  0  0
 100  0
 0  0160  60908  17896 36945200 0 0 2109   197  0  0
 100  0
 0  0160  62292  17916 36945200 0   432 2134   324  0  1
 99  0
 0  0160  62292  17916 36945200 0 0 2107   190  0  0
 100  0
 0  0160  62292  17916 36945200 0 0 2111   200  0  0
 100  0
 0  0160  62356  17916 36945200 0 0 2106   195  0  0
 100  0
 0  0160  62372  17916 36945200 0 0 2107   190  0  0
 100  0
 
 
 Wolfgang Zweimueller wrote:
 Hi All,

 I have read many posts about problems with Asterisk on some systems. I
 also set up Asterisk on many different boxes. But I have never seen
 the following...

 There is an IBM X346 (3.4GHz Xeon) with one Sangoma A104. This system
 is currently idle, that means there is nothing running except Asterisk
 (1.2.7.1). We are handling no calls now, but if I do a vmstat, I get
 peaks in system load up to 40%! Here is an example:

 procs ---memory-- ---swap-- -io --system--
 cpu
  r  b   swpd   free   buff  cache   si   sobibo   incs us
 sy id wa
  0  0  0 375080 161660 14223200 0 0 4160   187  0 
 4 96  0
  0  0  0 375080 161660 14223200 0 0 4251   207  0 
 1 98  0
  0  0  0 375080 161660 14223200 0 0 4205   179  0 
 9 92  0
  0  0  0 375080 161660 14223200 036 4151   217  0 
 3 97  0
  0  0  0 375080 161660 14223200 0 0 4026   187  0 
 0 100  0
  0  0  0 375080 161660 14223200 0 0 4042   205  0
 14 86  0
  0  0  0 375080 161660 14223200 0 0 4019   184  0
 38 63  0
  0  0  0 375080 161660 14223200 0 0 4062   208  0 
 0 100  0
  0  0  0 375080 161660 14223200 0 0 4028   196  0 
 2 99  0
  0  0  0 375080 161660 14223200 016 4075   223  0
 19 81  0
  1  0  0 375080 161660 14223200 0 0 4029   197  0 
 0 100  0
  0  0  0 375080 161660 14223200 0 0 4043   199  0 
 1 99  0
  0  0  0 375080 161660 14223200 0 0 4045   194  0 
 6 94  0
  0  0  0 375080 161660 14223200 0 0 4032   196  0
 24 77  0
  0  0  0 375080 161660 142232

[Asterisk-Users] Not joining queue when empty

2006-05-19 Thread Matt


Hi,
I have the following configured for my queue.   However, it seems that
because I have 'memberAgents' setup people still join the queue even
when no one is logged in!How can I have agents assigned like this,
yet still not allow joining the queue if they are not logged in?

[201]
wrapuptime=60
timeout=30
strategy=leastrecent
retry=15
queue-youarenext=queue-youarenext
queue-thereare=queue-thereare
queue-thankyou=queue-thankyou
queue-callswaiting=queue-callswaiting
music=default
monitor-join=yes
monitor-format=wav
member=Agent/1000
member=Agent/1001
member=Agent/1006
member=Agent/1011
member=Agent/1012
member=Agent/1022
member=Agent/1036
member=Agent/1016
member=Agent/1014
member=Agent/1045
member=Agent/1063
member=Agent/1231
maxlen=0
leavewhenempty=no
joinempty=No
context=
announce-holdtime=yes
announce-frequency=60
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Re: [Asterisk-Users] AsteriskOUT

2006-05-19 Thread Alex Robar

Hmm, I don't know... A user just looking for a DID or termination might not be subscribed to the business list. It's not like he's asking for services, he's just putting out a warning that they're not a great provider. The recent messages about NuPhone going down are related to the NuPhone business, but no one seems to have tagged them as off topic.
AlexOn 5/19/06, Ronald Wiplinger [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote: Hi all, Has anyone in the group tried the services of www.asteriskout.com. (lunaphone)
 Just thought of letting you all be aware not to fall on their services as it always seen attractive but to my experience they had always been pulling the the legs of customers with an approach of making money
 through ways that are thought to be non industrial.Are you talking about BUSINESS If so, there is another mailing list available!If it is a USERS question, and allow me please to quote the name of
this email list:Asterisk Users Mailing List - Non-Commercial Discussionthan you may specify your problem so that we can try to help you. Just wanted to warn you guys
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Re: [Asterisk-Users] R2/MFC Configuration.

2006-05-19 Thread Fernando Lujan


Moises Silva wrote:

Fernando: There are few or no people that will give you an Answer with
that information. The list is usually for people that already have
tried something and is experimienting some kind of specific problem.
Your question seems like ahhh it does not work, help me!. As far as
I can see you have 2 options. Please search in google for information
about how to configure, try something and then come back with a more
meaningfull question, or 2, hire some Asterisk consultant to make it
work.

Do you know Unicall?

please check http://www.voip-info.org/wiki/index.php?page=Asterisk+MFC+R2


Hi Moises,


I found this url later. I already download and install following the 
instructions in the URL above. The problem is that I can use the 
testcall program.


I don't know how to set up a configuration file. :( I try to create a 
testcall.conf file without success. So I stopped.


Thanks in advance.

Fernando Lujan
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Re: [Asterisk-Users] Polycom 601 -- programming buttons.


Hi Ken -


Hi, all.  I want to have a button on my receptionist's 601 that, when
pressed, will forward her current call to a given extension.  Is there any
way to do that?  I've tried to RTFM, but I'm coming up empty.


There's a couple of ways I can think of that would get this done.

1) For true one-button press for the user, you can map a speed dial to
a line appearance button.  On the asterisk end you'd have to handle
the forward in the dial plan using dbput/dbget and a variable (if
variable is on, dial the forward number, if variable is off, dial
the normal number).  I do something like this to enable a nightring
feature.

2) Just use the forward feature on the polycom.  The only disadvantage
is that it's two button presses for the user rather than one.  The
first time when you set it up, you'll have to put in the extension,
but after that you can enable/disable the forward by pressing the same
softkey twice.  This is probably the better answer, and is much easier
for you (requires firmware 1.5.x or later).


- Noah



On 5/18/06, Ken D'Ambrosio [EMAIL PROTECTED] wrote:

Hi, all.  I want to have a button on my receptionist's 601 that, when
pressed, will forward her current call to a given extension.  Is there any
way to do that?  I've tried to RTFM, but I'm coming up empty.

Thanks,

-Ken D'Ambrosio

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[Asterisk-Users] SpanDSP issues (oh fun!)

I am having classic frame slip symptoms on 4 different systems
with 4 different providers (all full PRIs, Qwest, XO, Xspedius, and First
Digital are the providers). By classic I mean pages cut short. I do not hear
clicks on calls however, and faxing from a fax machine plugged into the asterisk
box via an fxs port (no VoIP) faxes just fine through the PRI. Also, faxing
directly from said fxs port to spandsp (IE not traversing the PRI) works
wonderfully. I do not see any missed interupts on any of these systems,
the digium cards (tdm400's for fxs ports, and te110p's for PRI termination) are
not sharing IRQs with anything else, and they are not on the same IRQ themselves
IE, they are totally isolated in the IRQ space. Generally shorter faxes
(IE 3pages) get through just fine, however anything longer than that will
have a failed page about 50% of the time, and anything over 10 pages always has
a failed page (except on Xspedius, which has about a 50% fail rate above 10
pages). I have searched everywhere, and the answer to this problem seems to be frame slips on the PRI however, all 4 systems are set to sync with
the PSTN connection, I'm not running X on any of these systems, (although
turning X on doesn't negatively effect them either... IE I get the same number
of errors with X running or not). Also, there is no mention anywhere of a fix or even anything to try to reduce or eliminate frame slips.I recompiled spandsp to log the audio files, does anyone know what type of file that is? can I listen to it? Or is Steve the only one who knows how?
how come a standard fax machine can fax over
this PRI connection just fine, but spandsp cannot use it? If frame slips are
occurring, shouldn't they break faxing on the machine as well? If not, the machines
must be doing some error checking or something that spandsp does not currently
support so that they can deal with frame slips. Does spandsp do any sort of
error checking/page resend requesting? What would it take to make spandsp more
robust in this regard? Or is it just something that can't be overcome in
software, but that hardware modems in fax machines can handle? I am more than
willing to help out any way I can (including code, although I'd need a little
help getting up to speed on the code) to get spandsp working on these boxes.

Is there anything else to try? Basically all of the posts I've
been able to find say Don't share IRQs, turn off X, and make sure the PRI is
configured properly, well I've done all of that. Do I just have 4 bad PRIs
that can't keep time? These systems are deployed as a test solution for fax-email, they are the only 4 I've setup to use spandsp so 100% of the systems I have installed show this problem, and they
show it across telecom vendors all other things being equal (exact same
hardware, exact same OS, exact same asterisk, spandsp, etc versions asterisk
1.2.4, zaptel 1.2.4, libpri 1.2.2, spandsp 0.0.2pre25). I have tried later
versions of asterisk, zaptel, and libpri, but not a 0.0.3 version of spandsp. I
have sent faxes from many different fax machines, all show the problem.
I have attempted to run a utility I found
called ztclock... suprisingly of the 4 pri's the best as far as spandsp
results are concerned is Xspedius (of the 4 it is the only one that ever
receives 10+page faxes without error, still only about 50% of the time, but the
other 3 never receive more than 10 pages without failing a page). However,
according to ztclock it is the worst as far as frame slips are concerned. Here
are the results of Qwest vs Xspedius. Qwest PRI:ztclock - clock
source accuracy test (3 passes)Flushing input buffer...Flush
Complete.Test is approximately 3 minutes. Please wait...483328
samples in 60.416021 sec. (483329 sample intervals) 99.999794%483328 samples
in

60.416041 sec. (483329 sample intervals) 99.999794%483328 samples in
60.416097 sec. (483329 sample intervals) 99.999794%Estimate 8 frame
slips every 483.328003 seconds.Xspedius PRI:ztclock -
clock source accuracy test (3 passes)Flushing input buffer...Flush
Complete.Test is approximately 3 minutes. Please wait...483328
samples in 60.416018 sec. (483329 sample intervals) 99.999794%483328 samples
in 60.416037 sec. (483329 sample intervals) 99.999794%483328 samples in
60.514997 sec. (484120 sample intervals) 99.836136%Estimate 8 frame
slips every 0.610263 seconds.The other 2 PRIs show ztclock results
similar to Qwest, 8 frame slips every 400-500 seconds. Any help or
suggestions would be greatly appreciated. Thanks for your time,
Tom Christensen
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Re: [Asterisk-Users] Snom firmwares suck

2006-05-19 Thread Remco Barende




The displays are dying quite often, and firmware is buggy. I have tried
every firmware from 4.5 up to 5.x and 6.04 but keep having problems with
phones locking up or rebooting during an ongoing conversation.


Yes, as someone asked earlier on the list the displays do die at some 
random moment without any apparent reason



REALLY annoying for a phone that is advertised / targeted as a business
class phone


Hmmm... A random statement out of the blue... I assume that you meant
to add Does any kind soul have a suggestion to help out? :-)


Heh :)   It is pure frustration



I find that the snom phones can be over-sensetive to network glitches,
which with the default configuration can cause a reboot (usually
caused by cheap switches). Try changing the reboot on ethernet unplug
setting to ignore.


Good idea, I will change the settings to ignore.  The switches are 3Com 
gbit switches. Not sure if that would qualify as cheap :)


I have about 40 Snom 360's and I experienced this problem on my phone at 
home and some at the office using different firmware versions.


Thanks!
Remco

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Re: [Asterisk-Users] Snom firmwares suck

2006-05-19 Thread Dave Cotton

On Fri, 2006-05-19 at 14:19 +0100, Steve Davies wrote:
 On 5/19/06, Remco Barende [EMAIL PROTECTED] wrote:
  Most people seem quite positive about Snom phones, I cannot share this
  opinion.
 
  The displays are dying quite often, and firmware is buggy. I have tried
  every firmware from 4.5 up to 5.x and 6.04 but keep having problems with
  phones locking up or rebooting during an ongoing conversation.
 
  REALLY annoying for a phone that is advertised / targeted as a business
  class phone
 
 Hmmm... A random statement out of the blue... I assume that you meant
 to add Does any kind soul have a suggestion to help out? :-)
 

Old English saying A bad workman always blames his tools

Only problems I've had with Snoms have been Id10t user problems.


-- 
Dave Cotton [EMAIL PROTECTED]

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Re: [Asterisk-Users] Snom firmwares suck

2006-05-19 Thread Steve Davies


On 5/19/06, Remco Barende [EMAIL PROTECTED] wrote:

 I find that the snom phones can be over-sensetive to network glitches,
 which with the default configuration can cause a reboot (usually
 caused by cheap switches). Try changing the reboot on ethernet unplug
 setting to ignore.

Good idea, I will change the settings to ignore.  The switches are 3Com
gbit switches. Not sure if that would qualify as cheap :)

I have about 40 Snom 360's and I experienced this problem on my phone at
home and some at the office using different firmware versions.


I would be interested to know whether this works for you, just as an
extra datapoint.

Regards,
Steve
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Re: [Asterisk-Users] Not joining queue when empty

2006-05-19 Thread Time Bandit


I have the following configured for my queue.   However, it seems that
because I have 'memberAgents' setup people still join the queue even
when no one is logged in!How can I have agents assigned like this,
yet still not allow joining the queue if they are not logged in?


I think you have to use

joinempty = strict
leavewhenempty = strict

hth
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Re: [Asterisk-Users] Snom firmwares suck

2006-05-19 Thread Dr. Michael J. Chudobiak


Old English saying A bad workman always blames his tools


I don't think that's fair... these are very complicated phones, made in 
China for very low prices. Problems do occur with them.


Some Snom LCDs do have problems.

There are firmware glitches, though I've only run into minor ones.

Overall though, they are very good phones.


- Mike
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[Asterisk-Users] RTP Packetization

2006-05-19 Thread Patrick Neubauer


Hi all,

I need to be able to adjust packet sizes and found the patch at 
http://bugs.digium.com/view.php?id=5162


Thus, I checked out and compiled 
http://svn.digium.com/view/asterisk-old/team/group/5162_rtp_packetization


I added the line packetization = 30 for one peer in my sip.conf and 
started asterisk with the -I switch for async RTP.
That's all it takes according to the 5162 issue page. Nevertheless, 
asterisk still keeps sending it 20ms packets, even though a sip show 
peer foobar shows Packetization: 30.


What could be wrong? What about that ztdummy thing for internal timing? 
Is this necessary to run asterisk properly? Is it important for 
packetization?


Regards, Patrick
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Re: [Asterisk-Users] Experience with IBM X346 machines and Sangoma

2006-05-19 Thread Cosmin Prund

I wanted to see where those periodical spikes are coming from so I 
started shutting things down. The first thing to go was Asterisk. Next 
went many different things like mysql and ntpd. Finally I killed zaptel 
(/etc/init.d/zaptel stop) - and the spiking stoped!


Next I rebooted and I've done /etc/init.d/zaptel stop straight away. 
The spiking stoped again. I've done /etc/init.d/zaptel start and 
spiking started again!


Is there something funny happening with my zaptel?
Wolfgang Zweimueller, can you give this a try too? Does your spiking 
stop when you stop zaptel?

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Re: [Asterisk-Users] just softphone

2006-05-19 Thread Ralph Liebessohn

On 5/18/06, Stefan Märkle [EMAIL PROTECTED] wrote:
Try puting apermit=0.0.0.0/0.0.0.0In the sip.conf for your two phones.BTW: your extensions.conf looks silly, you'll only be able to call test3 from test3.Busy most of the time ;-)
Stefan MärkleTry puting apermit=0.0.0.0/0.0.0.0in the sip.conf for your two phones.Fine, I tried it.But doesn't solve. So I just started from zero and installed all the system again and it starts to work almost normally.
BTW: your extensions.conf looks silly, you'll only be able to call test3 from test3.Busy most of the time ;-)That's for the times when I feel alone and wants to talk to myself ! ! !It was just for tests.
-- Ralph LiebessohnICQ: 74835911Skype: liebessohn
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Re: [Asterisk-Users] regexp


Hi Domenico -


 First, you can remove the quotes aorund your variable
 reference.  I've seen examples with it, but you don't need
 it.
I'm not sure: if variable is empty, you got an error.
In addition, double quotes around text that may contain spaces
will force the surrounded text to be evaluated as a single token.


Ah.  Good point.  I guess it is a good idea to use quotes.



 Second, I'm not sure what the tilde does after the equal
 sign, but asterisk won't understand it.
What?
':' and '=~' are regexp operators in Asterisk.


Ah, again.  Undocumented functionality.  I looked through the source
code for ast_expr2.c, and you are so correct.  Still, in this
application, I think you can just use a single equal sign.


- Noah




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Re: [Asterisk-Users] Not joining queue when empty

2006-05-19 Thread Matt


AHA!  That did it... And to think I just setup a bunch of time based
routing :)  Oh well guess that keeps people from getting into the
queue if an agent forgets to logout!

On 5/19/06, Time Bandit [EMAIL PROTECTED] wrote:

 I have the following configured for my queue.   However, it seems that
 because I have 'memberAgents' setup people still join the queue even
 when no one is logged in!How can I have agents assigned like this,
 yet still not allow joining the queue if they are not logged in?

I think you have to use

joinempty = strict
leavewhenempty = strict

hth
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Re: [Asterisk-Users] Watchguard Firebox 1000 woes

2006-05-19 Thread RandyW





Greetings Kerry,

I've tinkered quite a bit with Firebox's (Specifically Firebox III's),
and may be of some value. What version of the system manager are you
using? What version of the Firebox are you using?? (I'm taking that
the "1000" is the Firebox III 1000, am I right?)

What type of logging do you have running? Syslog? Event Processor?

Would you mind sharing your config file?

Would you mind sharing what type of SIP connection you are trying to
accomplish. I too have had significant challenges with WG lately, so I
can see them saying that.

I'm not sure what I can do, but I'd be glad to help.

RandyW

Kerry Garrison wrote:

  
  
  We
are trying to setup a sip connection behind a Watchguard Firebox 1000
and it is simply put...not working. The ports are all forwarded but the
packets are not going out. It is as if the firewall simply ignores SIP
packets. Has anyone seen this or have any idea what the issue could be?
Watchguard so far has been of zero help.
  
  
  Kerry Garrison
Director of Technical Services
  Tech Data Pros - Orange County's Mobile IT Service
Provider
  (949)502-7819 x200- [EMAIL PROTECTED]
  http://www.techdatapros.com
  
  

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Re: [Asterisk-Users] Snom firmwares suck

2006-05-19 Thread Dave Cotton

On Fri, 2006-05-19 at 10:21 -0400, Dr. Michael J. Chudobiak wrote:
  Old English saying A bad workman always blames his tools
 
 I don't think that's fair... these are very complicated phones, made in 
 China for very low prices. Problems do occur with them.
 
 Some Snom LCDs do have problems.
 
 There are firmware glitches, though I've only run into minor ones.
 
 Overall though, they are very good phones.

And that is the whole point of the saying.

I object to the type of posting that says XYZ sucks it is the language
and attitude of children. As Steve Davies said I assume that you meant
to add Does any kind soul have a suggestion to help out? :-)

-- 
Dave Cotton [EMAIL PROTECTED]

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RE: [Asterisk-Users] Snom firmwares suck

2006-05-19 Thread The VoIP Connection

I can confirm that the issue with the display is usually a hardware problem,
not firmware.  If the phone has this issue, it may be related to the other
issues (locking up, etc.).  I suggest you return the phones that have this
problem for warrantee exchange.

As far as the firmware goes, the production versions generally have a few
minor issues but are pretty sound overall (relative to other vendors). -Mike

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

 -Original Message-
 From: Dr. Michael J. Chudobiak [mailto:[EMAIL PROTECTED] 
 Sent: Friday, May 19, 2006 9:21 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Cc: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Snom firmwares suck
 
 Remco Barende wrote:
  Most people seem quite positive about Snom phones, I cannot 
 share this 
  opinion.
  
  The displays are dying quite often, and firmware is buggy. I have 
  tried every firmware from 4.5 up to 5.x and 6.04 but keep having 
  problems with phones locking up or rebooting during an 
 ongoing conversation.
  
  REALLY annoying for a phone that is advertised / targeted as a 
  business class phone
 
 Remco,
 
 I have a dozen Snom 360s. One had a defective LCD that would 
 become garbled after time. Snom support quickly confirmed 
 that this was a known issue, and my vendor (voipsupply) 
 quickly sent a replacement.
 
 I've never seen any lockups or reboots. I reboot the phones 
 each night at midnight, just to be safe - try doing that to 
 see if it reduces problems. I've posted a sample perl cron 
 script at 
 http://www.voip-info.org/wiki/view/Asterisk+phone+snom to show how.
 
 I use shielded ethernet cables (STP) everywhere too. Try 
 that - good grounding may be beneficial. It can't hurt, anyway.
 
 Snom support is pretty responsive. Try emailing 
 [EMAIL PROTECTED]; they have fixed some issues for me (for 
 example, the clock was showing the wrong time due to daylight 
 savings time problems).
 
 Try using a Grandstream GXP-2000 phone, and you'll see why 
 people like the Snoms :-)
 
 Hope this helps - let us know if anything makes a difference!
 
 
 - Mike
 
 

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RE: [Asterisk-Users] FAX over PRI

2006-05-19 Thread Steve Hanselman

Sorry for the late reply but both of these are fine, we use spandsp to
print some faxes and email others.

We also route via a PRI to our other phone system to hylafax on an
analog modem and also to an analog fax.

So what you want to do is fine and will work.

Steve


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Gaudette
Sent: 21 March 2006 20:34
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] FAX over PRI

Hmmm, Im not so sure I can apply this to me though.  I just want to do
Fax-To-Email using PRI channels as the incoming lines.  Not so much
transfer
to a real fax.

I am assuming that this is easily done with Asterisk? (I did it before
with
Asterisk SIP, but it only worked once every 10 tries or so)

Mike

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: March 21, 2006 3:25 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] FAX over PRI

On Tuesday 21 March 2006 15:09, Michael Gaudette wrote:
 How should I consider Fax over PRI channels with Asterisk?  Is the
 quality and reliability good, or should I be prepared for alot of
grief?

I'm having good success doing fax over PRI using a TE405; one span to
the
PRI, the other to an FXS channel bank that is almost obscenely
underutilized
(3 channels).

I also have channel bank - T100P - IAX2 - TE405 - PRI, where the IAX2
link
is a 1-hop SDSL (VOIP only) data link.  This works well too.

-A.
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Re: [Asterisk-Users] Snom firmwares suck

2006-05-19 Thread RandyW





I'm gonna jump in here and join Dave here.

It simply is of no value to just SPAM your opinion around without
either providing real facts or asking for assistance. We are here to
ask for or give out help. This just isn't productive.

Now, if you want a detailed, painfully so, explanation why Sun's DLT
8000 tape drives are "underperformers" (notice I didn't "vomitous
garbage") I would be happy to help you out. Off-line of course... 

grin

RandyW

Dave Cotton wrote:

  On Fri, 2006-05-19 at 10:21 -0400, Dr. Michael J. Chudobiak wrote:
  
  

  Old English saying "A bad workman always blames his tools"
  

I don't think that's fair... these are very complicated phones, made in 
China for very low prices. Problems do occur with them.

Some Snom LCDs do have problems.

There are firmware glitches, though I've only run into minor ones.

Overall though, they are very good phones.

  
  
And that is the whole point of the saying.

I object to the type of posting that says "XYZ sucks" it is the language
and attitude of children. As Steve Davies said "I assume that you meant
to add "Does any kind soul have a suggestion to help out?" :-)"

  



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Re: [Asterisk-Users] Polycom 601 -- programming buttons.

2006-05-19 Thread Jerry Jones

Uh - If the OP is trying to transfer an existing call, then he should  
be using transfer not forward. You may not access the forward  
function on a polycom with an active call. Forward will send  
subsequent calls to your specified destination, not existing calls.



On May 19, 2006, at 8:53 AM, Noah Miller wrote:


Hi Ken -


Hi, all.  I want to have a button on my receptionist's 601 that, when
pressed, will forward her current call to a given extension.  Is  
there any

way to do that?  I've tried to RTFM, but I'm coming up empty.


There's a couple of ways I can think of that would get this done.

1) For true one-button press for the user, you can map a speed dial to
a line appearance button.  On the asterisk end you'd have to handle
the forward in the dial plan using dbput/dbget and a variable (if
variable is on, dial the forward number, if variable is off, dial
the normal number).  I do something like this to enable a nightring
feature.

2) Just use the forward feature on the polycom.  The only disadvantage
is that it's two button presses for the user rather than one.  The
first time when you set it up, you'll have to put in the extension,
but after that you can enable/disable the forward by pressing the same
softkey twice.  This is probably the better answer, and is much easier
for you (requires firmware 1.5.x or later).


- Noah



On 5/18/06, Ken D'Ambrosio [EMAIL PROTECTED] wrote:

Hi, all.  I want to have a button on my receptionist's 601 that, when
pressed, will forward her current call to a given extension.  Is  
there any

way to do that?  I've tried to RTFM, but I'm coming up empty.

Thanks,

-Ken D'Ambrosio

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Re: [Asterisk-Users] Digium card firmware

2006-05-19 Thread Kevin P. Fleming

Bruno de Assumpção Loureiro wrote:

 how can I know which version my TE405p Digium card is?

It will be reported in the kernel message log (dmesg) when you load the
driver and it binds to the card.
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[Asterisk-Users] IAX Trunk

2006-05-19 Thread Forrest Beck









I have two servers setup on different locations. They are
both setup as peers and users to each other.



Server 1 iax.conf:

[ms-to-us]

type=user

username=ms-to-us

secret=ms-to-us

context=upperschool

[us-to-ms]

type=peer

username=us-to-ms

secret=us-to-ms

host=10.11.1.112

trunk=yes



Server 2 iax.conf:

[ms-to-us]

type=peer

username=ms-to-us

secret=ms-to-us

host=10.11.1.111

trunk=yes

[us-to-ms]

type=user

username=us-to-ms

secret=us-to-ms

context=middleschool



All works great. But I am curious. 



Senario: If a call is initiated from Server 1 to Server 2, a
trunk is established. While that call is progress another call is established from
Server 2 to Server 1. Is a new trunk created, or is the same one used?






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[Asterisk-Users] Non automated call parking

2006-05-19 Thread Steven




From 
discussions with the receptionist staff, this is what we need:X number 
of buttons for parking slots.These buttons should be lit when a call is 
parked there.When on a call, just pushing an unlit button will park the 
call. (The do not want to hit hold, transfer, etc.)Hitting Transfer and the 
button would most likely be OK.To pick up a call, they can push the parking 
slot button.The parking slot should be a dailable extension, so that 
everyone else can just dial 5401, etc to pick up the call.I assume that 
there is some kind of local channel I can do this with and avoid any of the 
automated parking systems like Park,ParkandAnnounce or ValetParking, all of 
which have their pros and cons.Digging through res_features.c it is hard 
to discern where the call really is when parked (what kind of channel it is) so 
I am notsure if it can be reproduced in dialplan.Has anyone done 
manual call parking this 
way?Steven
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Re: [Asterisk-Users] Quad BRI card

2006-05-19 Thread Craig Guy

By the last sentence I mean that only the person or company holding the 
A-tick can put the sticker on the cards. Paralell importation refers to 
'grey' imports that don't come through the vendors sanctioned distribution 
channels.  For example I know that the fritz! has passed approval because 
this guy has gone through the approval process.  The Australian distributor 
sells them for $400, I can get them off eBay in Europe for $20 per card - 
the exact same card.  $400 is just pure extortion and is going a hell of a 
long way to prevent the adoption of Asterisk in this country where BRI is 
the norm and PRI is outrageously expensive.


If I had a spare $20k or so then I'd approve the card myself and sell them 
at a more realistic price.


Craig

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

Sent: Friday, May 19, 2006 8:54 AM
Subject: Re: [Asterisk-Users] Quad BRI card


On 5/18/06, Craig Guy [EMAIL PROTECTED] wrote:

Any device to legally connect to the PSTN in Australia must be approved by
the regulatory body.  A process that usually costs at least $20,000 and 
only
allows the permit holder to sell the product for conneciton to the pstn. 
It

is a very high barrier to entry for the Australian market.  There is a guy
in Victoria who certified the Fritz! card and charges $400 each for them.
Paralell imports are not allowed to be connected.


Ah, so that's why they're so expensive :(

Sorry, what do you mean by that last sentence?

Andrew

--
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
 -- Bill Garrett
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Re: [Asterisk-Users] Polycom 601 -- programming buttons.


Hi Ken, Jerry -


 Hi, all.  I want to have a button on my receptionist's 601 that, when
 pressed, will forward her current call to a given extension.  Is
 there any
 way to do that?  I've tried to RTFM, but I'm coming up empty.

Uh - If the OP is trying to transfer an existing call, then he should
be using transfer not forward. You may not access the forward
function on a polycom with an active call. Forward will send
subsequent calls to your specified destination, not existing calls.


Oh, good call, Jerry.  Ken, on second inspection, your wording does
kind of read like you want to transfer a call rather than forward it.
If that's the case, you can do something like the 1st of my
suggestions.  Here's what I do:

1. Enable Blind Transfers in asterisk features.conf (and in your
dialplan), and map to '#'
2. In sip.cfg, remap the Transfer Hard Key to be '#' like this:
   keys   key.IP_600.37.function.prim=DialpadPound /
3. Map a line key to be a speed dial for the extension you want (see manual).

Then, to actually use it, just press the transfer hard key and the
line appearance that's your speed dial.  It's two button presses, but
I think that's the closest you can come to a single button transfer to
another extension with a Polycom.

- Noah



On 5/19/06, Jerry Jones [EMAIL PROTECTED] wrote:

Uh - If the OP is trying to transfer an existing call, then he should
be using transfer not forward. You may not access the forward
function on a polycom with an active call. Forward will send
subsequent calls to your specified destination, not existing calls.


On May 19, 2006, at 8:53 AM, Noah Miller wrote:

 Hi Ken -

 Hi, all.  I want to have a button on my receptionist's 601 that, when
 pressed, will forward her current call to a given extension.  Is
 there any
 way to do that?  I've tried to RTFM, but I'm coming up empty.

 There's a couple of ways I can think of that would get this done.

 1) For true one-button press for the user, you can map a speed dial to
 a line appearance button.  On the asterisk end you'd have to handle
 the forward in the dial plan using dbput/dbget and a variable (if
 variable is on, dial the forward number, if variable is off, dial
 the normal number).  I do something like this to enable a nightring
 feature.

 2) Just use the forward feature on the polycom.  The only disadvantage
 is that it's two button presses for the user rather than one.  The
 first time when you set it up, you'll have to put in the extension,
 but after that you can enable/disable the forward by pressing the same
 softkey twice.  This is probably the better answer, and is much easier
 for you (requires firmware 1.5.x or later).


 - Noah



 On 5/18/06, Ken D'Ambrosio [EMAIL PROTECTED] wrote:
 Hi, all.  I want to have a button on my receptionist's 601 that, when
 pressed, will forward her current call to a given extension.  Is
 there any
 way to do that?  I've tried to RTFM, but I'm coming up empty.

 Thanks,

 -Ken D'Ambrosio

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Re: [Asterisk-Users] Non automated call parking


Hi Steven -

It's not in the general release yet and it doesn't do everything you
want, but check out the metermaid patch
(http://bugs.digium.com/view.php?id=5779).

If you're interested, I also wrote a patch to do one button parking
(http://bugs.digium.com/view.php?id=6340).  You can use it now, but as
the bugs says, it will not be included in future releases as the
applicationmap part of res_features.c will eventually be fixed to do
this correctly.


- Noah



On 5/19/06, Steven [EMAIL PROTECTED] wrote:



From discussions with the receptionist staff, this is what we need:

X number of buttons for parking slots.
These buttons should be lit when a call is parked there.
When on a call, just pushing an unlit button will park the call. (The do not
want to hit hold, transfer, etc.)
Hitting Transfer and the button would most likely be OK.
To pick up a call, they can push the parking slot button.
The parking slot should be a dailable extension, so that everyone else can
just dial 5401, etc to pick up the call.

I assume that there is some kind of local channel I can do this with and
avoid any of the automated parking systems like Park,
ParkandAnnounce or ValetParking, all of which have their pros and cons.

Digging through res_features.c it is hard to discern where the call really
is when parked (what kind of channel it is) so I am not
sure if it can be reproduced in dialplan.

Has anyone done manual call parking this way?


Steven

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[Asterisk-Users] British English voice files are ready for download

2006-05-19 Thread Mark Phillips

Hi folks,

With thanks to Alison Keenan (another Alison!) for the voice, Chris
Bagnal for converting from 44k wav to sln and finally Terje Elde for
debugging my HTML code, the British English files are now ready for
download.

They can be got from http://www.enicomms.com/cutglassivr/ 

Thanks and don't forget to practice safe IAX ;-}

Mark

-- 
Mark Phillips [EMAIL PROTECTED]

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Re: [Asterisk-Users] FAX over PRI

I have had nothing but problems receiving faxes over PRIs with spandsp. I currently have 4 systems, 4 PRIs from 4 different providers... none of them get better than 50% success rates receiving faxes in spandsp, I constantly get cut off pages. No body seems to have a fix for it, and it is really frustrating. Supposedly it is caused by frame slips on the PRI, but if that is the case, I am 4 for 4 getting crappy PRIs that can't keep time.
These same boxes work fine when receiving faxes over fxo ports, or if I plug a fax machine into an fxs port and call in to a spandsp extension the fax will be received just fine, so I am left thinking it must be the PRIs, but if all PRIs are this bad, how can anybody be using them?
TomOn 5/19/06, Steve Hanselman [EMAIL PROTECTED] wrote:
Sorry for the late reply but both of these are fine, we use spandsp toprint some faxes and email others.We also route via a PRI to our other phone system to hylafax on ananalog modem and also to an analog fax.
So what you want to do is fine and will work.Steve-Original Message-From: [EMAIL PROTECTED][mailto:
[EMAIL PROTECTED]] On Behalf Of MichaelGaudetteSent: 21 March 2006 20:34To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] FAX over PRIHmmm, Im not so sure I can apply this to me though.I just want to doFax-To-Email using PRI channels as the incoming lines.Not so muchtransferto a real fax.
I am assuming that this is easily done with Asterisk? (I did it beforewithAsterisk SIP, but it only worked once every 10 tries or so)Mike-Original Message-From: 
[EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of AndrewKohlsmithSent: March 21, 2006 3:25 PM
To: asterisk-users@lists.digium.comSubject: Re: [Asterisk-Users] FAX over PRIOn Tuesday 21 March 2006 15:09, Michael Gaudette wrote: How should I consider Fax over PRI channels with Asterisk?Is the
 quality and reliability good, or should I be prepared for alot ofgrief?I'm having good success doing fax over PRI using a TE405; one span tothePRI, the other to an FXS channel bank that is almost obscenely
underutilized(3 channels).I also have channel bank - T100P - IAX2 - TE405 - PRI, where the IAX2linkis a 1-hop SDSL (VOIP only) data link.This works well too.-A.___
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of the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat any review, distribution or copying of this document is strictly prohibited. If you have
receivedthis communication in error, please notify Brendata immediately on:+44 (0)1268 466100, or email '[EMAIL PROTECTED]'Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BXUK
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Re: [Asterisk-Users] FAX over PRI


Hi Tom -


I have had nothing but problems receiving faxes over PRIs with spandsp.  I
currently have 4 systems, 4 PRIs from 4 different providers... none of them
get better than 50% success rates receiving faxes in spandsp, I constantly
get cut off pages.  No body seems to have a fix for it, and it is really
frustrating.  Supposedly it is caused by frame slips on the PRI, but if
that is the case, I am 4 for 4 getting crappy PRIs that can't keep time.


Just an aside thought (sorry to hijack the thread, Steve):

50% - Ouch.  I only have one PRI at one of our offices, but we use it
to receive faxes that are directly sent via Digium FXS to an analog
fax machine.  I've never formally tallied up the transmission errors,
but we get something close to 100%.  Maybe spandsp is an issue here.

- Noah



On 5/19/06, Tom Christensen [EMAIL PROTECTED] wrote:

I have had nothing but problems receiving faxes over PRIs with spandsp.  I
currently have 4 systems, 4 PRIs from 4 different providers... none of them
get better than 50% success rates receiving faxes in spandsp, I constantly
get cut off pages.  No body seems to have a fix for it, and it is really
frustrating.  Supposedly it is caused by frame slips on the PRI, but if
that is the case, I am 4 for 4 getting crappy PRIs that can't keep time.

These same boxes work fine when receiving faxes over fxo ports, or if I plug
a fax machine into an fxs port and call in to a spandsp extension the fax
will be received just fine, so I am left thinking it must be the PRIs, but
if all PRIs are this bad, how can anybody be using them?

Tom


On 5/19/06, Steve Hanselman [EMAIL PROTECTED] wrote:
 Sorry for the late reply but both of these are fine, we use spandsp to
 print some faxes and email others.

 We also route via a PRI to our other phone system to hylafax on an
 analog modem and also to an analog fax.

 So what you want to do is fine and will work.

 Steve


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto: [EMAIL PROTECTED] On
Behalf Of Michael
 Gaudette
 Sent: 21 March 2006 20:34
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] FAX over PRI

 Hmmm, Im not so sure I can apply this to me though.  I just want to do
 Fax-To-Email using PRI channels as the incoming lines.  Not so much
 transfer
 to a real fax.

 I am assuming that this is easily done with Asterisk? (I did it before
 with
 Asterisk SIP, but it only worked once every 10 tries or so)

 Mike

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On
Behalf Of Andrew
 Kohlsmith
 Sent: March 21, 2006 3:25 PM
 To: asterisk-users@lists.digium.com
 Subject: Re: [Asterisk-Users] FAX over PRI

 On Tuesday 21 March 2006 15:09, Michael Gaudette wrote:
  How should I consider Fax over PRI channels with Asterisk?  Is the
  quality and reliability good, or should I be prepared for alot of
 grief?

 I'm having good success doing fax over PRI using a TE405; one span to
 the
 PRI, the other to an FXS channel bank that is almost obscenely
 underutilized
 (3 channels).

 I also have channel bank - T100P - IAX2 - TE405 - PRI, where the IAX2
 link
 is a 1-hop SDSL (VOIP only) data link.  This works well too.

 -A.
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not the intended
 recipient then you are hereby notified that you have received this
document in error and
 that any review, distribution or copying of this document is strictly
prohibited. If you have
 received  this communication in error, please notify Brendata immediately
on:

 +44 (0)1268 466100, or email '[EMAIL PROTECTED]'

 Brendata (UK) Ltd
 Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX  UK
 Registered Office as above. Registered in England No. 2764339

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To

RE: [Asterisk-Users] FAX over PRI









I have the same problem, Switched to
HylaFax and IAXModem and had MUCH better luck. MUCH better being defined as not
a single usable fax to only missing about 1%. Not bad.



Spandsp or rather RXFax works on a few
machines I hae quite well but on others it does not work at all, I have had
good retruns with the HylaFax.IAXmodem combo on machines that could not use the
RxFax. 















From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Tom Christensen
Sent: Friday, May 19, 2006 12:18
PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] FAX
over PRI





I have had nothing but
problems receiving faxes over PRIs with spandsp. I currently have 4
systems, 4 PRIs from 4 different providers... none of them get better than 50%
success rates receiving faxes in spandsp, I constantly get cut off pages.
No body seems to have a fix for it, and it is really frustrating. Supposedly
it is caused by frame slips on the PRI, but if that is the case, I
am 4 for 4 getting crappy PRIs that can't keep time. 

These same boxes work fine when receiving faxes over fxo ports, or if I plug a
fax machine into an fxs port and call in to a spandsp extension the fax will be
received just fine, so I am left thinking it must be the PRIs, but if all PRIs
are this bad, how can anybody be using them? 

Tom



On 5/19/06, Steve
Hanselman [EMAIL PROTECTED]
wrote:

Sorry for the late reply but both of these are fine, we use spandsp to
print some faxes and email others.

We also route via a PRI to our other phone system to hylafax on an
analog modem and also to an analog fax. 

So what you want to do is fine and will work.

Steve


-Original Message-
From: [EMAIL PROTECTED]
[mailto: [EMAIL PROTECTED]]
On Behalf Of Michael
Gaudette
Sent: 21 March 2006 20:34
To: 'Asterisk Users Mailing List - Non-Commercial
 Discussion' 
Subject: RE: [Asterisk-Users] FAX over PRI

Hmmm, Im not so sure I can apply this to me though.I just want to
do
Fax-To-Email using PRI channels as the incoming lines.Not so much
transfer
to a real fax. 

I am assuming that this is easily done with Asterisk? (I did it before
with
Asterisk SIP, but it only worked once every 10 tries or so)

Mike

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]
On Behalf Of Andrew
Kohlsmith
Sent: March 21, 2006 3:25 PM 
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] FAX over PRI

On Tuesday 21 March 2006 15:09, Michael Gaudette wrote:
 How should I consider Fax over PRI channels with Asterisk?Is
the 
 quality and reliability good, or should I be prepared for alot of
grief?

I'm having good success doing fax over PRI using a TE405; one span to
the
PRI, the other to an FXS channel bank that is almost obscenely 
underutilized
(3 channels).

I also have channel bank - T100P - IAX2 - TE405 - PRI, where the IAX2
link
is a 1-hop SDSL (VOIP only) data link.This works well too.

-A.
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that any review, distribution or copying of this document is strictly
prohibited. If you have 
receivedthis communication in error, please notify Brendata
immediately on:

+44 (0)1268 466100, or email '[EMAIL PROTECTED]'

Brendata (UK)
Ltd
Nevendon Hall, Nevendon Road,
Basildon, Essex. SS13 1BXUK 
Registered Office as above. Registered in England No. 2764339

See our current vacancies at www.brendata.co.uk
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RE: [Asterisk-Users] Polycom 601 -- programming buttons.

2006-05-19 Thread Douglas Garstang

 Hi Ken, Jerry -
 
   Hi, all.  I want to have a button on my receptionist's 
 601 that, when
   pressed, will forward her current call to a given extension.  Is
   there any
   way to do that?  I've tried to RTFM, but I'm coming up empty.
 
  Uh - If the OP is trying to transfer an existing call, then 
 he should
  be using transfer not forward. You may not access the forward
  function on a polycom with an active call. Forward will send
  subsequent calls to your specified destination, not existing calls.

Not true. The polycom's will let you forward an incoming call, for that call 
only. When a new call comes in, the phone is not forwarded anymore. The phone 
sends a 'Moved Temporarily' back to Asterisk, which re-enters the dialplan as 
Local. If someone transfers the call, the phone sends a REFER to Asterisk, who 
sets rdnis and re-enters the dialplan again as Local.
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RE: [Asterisk-Users] FAX over PRI

2006-05-19 Thread Steve Hanselman









We get the occasional bad fax, but it
really is an occasional one, other than that, its fine.



We dont get any CRC errors or clock
slips on the PRI, Id certainly say that it would be a good starting
point to check the counters on these, Id also check that your drives are
using DMA depending on your hardware, we had a customer a while ago who ended
up doing a self install and none of his drives were enabled for DMA.



Steve













From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Christensen
Sent: 19 May 2006 17:18
To: Asterisk Users Mailing List -
 Non-Commercial Discussion
Subject: Re: [Asterisk-Users] FAX
over PRI





I have had nothing but
problems receiving faxes over PRIs with spandsp. I currently have 4
systems, 4 PRIs from 4 different providers... none of them get better than 50%
success rates receiving faxes in spandsp, I constantly get cut off pages.
No body seems to have a fix for it, and it is really frustrating.
Supposedly it is caused by frame slips on the PRI, but if that is
the case, I am 4 for 4 getting crappy PRIs that can't keep time. 

These same boxes work fine when receiving faxes over fxo ports, or if I plug a
fax machine into an fxs port and call in to a spandsp extension the fax will be
received just fine, so I am left thinking it must be the PRIs, but if all PRIs
are this bad, how can anybody be using them? 

Tom



On 5/19/06, Steve
 Hanselman [EMAIL PROTECTED]
wrote:

Sorry for the late reply but both of these are fine, we use spandsp to
print some faxes and email others.

We also route via a PRI to our other phone system to hylafax on an
analog modem and also to an analog fax. 

So what you want to do is fine and will work.

Steve


-Original Message-
From: [EMAIL PROTECTED]
[mailto: [EMAIL PROTECTED]]
On Behalf Of Michael
Gaudette
Sent: 21 March 2006 20:34
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
Subject: RE: [Asterisk-Users] FAX over PRI

Hmmm, Im not so sure I can apply this to me though.I just want to
do
Fax-To-Email using PRI channels as the incoming lines.Not so much
transfer
to a real fax. 

I am assuming that this is easily done with Asterisk? (I did it before
with
Asterisk SIP, but it only worked once every 10 tries or so)

Mike

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]
On Behalf Of Andrew
Kohlsmith
Sent: March 21, 2006 3:25 PM 
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] FAX over PRI

On Tuesday 21 March 2006 15:09, Michael Gaudette wrote:
 How should I consider Fax over PRI channels with Asterisk?Is
the 
 quality and reliability good, or should I be prepared for alot of
grief?

I'm having good success doing fax over PRI using a TE405; one span to
the
PRI, the other to an FXS channel bank that is almost obscenely 
underutilized
(3 channels).

I also have channel bank - T100P - IAX2 - TE405 - PRI, where the IAX2
link
is a 1-hop SDSL (VOIP only) data link.This works well too.

-A.
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of the addressee only. It may also be privileged information. If you are not
the intended
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error and
that any review, distribution or copying of this document is strictly
prohibited. If you have 
receivedthis communication in error, please notify Brendata
immediately on:

+44 (0)1268 466100, or email '[EMAIL PROTECTED]'

Brendata (UK) Ltd
Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BXUK 
Registered Office as above. Registered in England No. 2764339

See our current vacancies at www.brendata.co.uk
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The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat any review, distribution or copying of this document is strictly prohibited. If you have received  this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX  UKRegistered Office as above. Registered in England No.

RE: [Asterisk-Users] FAX over PRI

2006-05-19 Thread Colin Anderson

My failure rate, objectively measured, is 3.8%, and this is with 100 - 400 a
day. Other than clock slips (which definitely adversely affects fax) I also
note that load is an issue. A system with a higher load has a higher
probability of failing the fax. Unfortunately, I don't have precise numbers,
as I have gotten a feel for this by watching 2 SSH windows to the same box,
1 running top and the other running the Asterisk console. 



-Original Message-
From: Noah Miller [mailto:[EMAIL PROTECTED]
Sent: Friday, May 19, 2006 10:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] FAX over PRI


Hi Tom -

 I have had nothing but problems receiving faxes over PRIs with spandsp.  I
 currently have 4 systems, 4 PRIs from 4 different providers... none of
them
 get better than 50% success rates receiving faxes in spandsp, I constantly
 get cut off pages.  No body seems to have a fix for it, and it is really
 frustrating.  Supposedly it is caused by frame slips on the PRI, but if
 that is the case, I am 4 for 4 getting crappy PRIs that can't keep time.

Just an aside thought (sorry to hijack the thread, Steve):

50% - Ouch.  I only have one PRI at one of our offices, but we use it
to receive faxes that are directly sent via Digium FXS to an analog
fax machine.  I've never formally tallied up the transmission errors,
but we get something close to 100%.  Maybe spandsp is an issue here.

- Noah



On 5/19/06, Tom Christensen [EMAIL PROTECTED] wrote:
 I have had nothing but problems receiving faxes over PRIs with spandsp.  I
 currently have 4 systems, 4 PRIs from 4 different providers... none of
them
 get better than 50% success rates receiving faxes in spandsp, I constantly
 get cut off pages.  No body seems to have a fix for it, and it is really
 frustrating.  Supposedly it is caused by frame slips on the PRI, but if
 that is the case, I am 4 for 4 getting crappy PRIs that can't keep time.

 These same boxes work fine when receiving faxes over fxo ports, or if I
plug
 a fax machine into an fxs port and call in to a spandsp extension the fax
 will be received just fine, so I am left thinking it must be the PRIs, but
 if all PRIs are this bad, how can anybody be using them?

 Tom


 On 5/19/06, Steve Hanselman [EMAIL PROTECTED] wrote:
  Sorry for the late reply but both of these are fine, we use spandsp to
  print some faxes and email others.
 
  We also route via a PRI to our other phone system to hylafax on an
  analog modem and also to an analog fax.
 
  So what you want to do is fine and will work.
 
  Steve
 
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto: [EMAIL PROTECTED] On
 Behalf Of Michael
  Gaudette
  Sent: 21 March 2006 20:34
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
  Subject: RE: [Asterisk-Users] FAX over PRI
 
  Hmmm, Im not so sure I can apply this to me though.  I just want to do
  Fax-To-Email using PRI channels as the incoming lines.  Not so much
  transfer
  to a real fax.
 
  I am assuming that this is easily done with Asterisk? (I did it before
  with
  Asterisk SIP, but it only worked once every 10 tries or so)
 
  Mike
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On
 Behalf Of Andrew
  Kohlsmith
  Sent: March 21, 2006 3:25 PM
  To: asterisk-users@lists.digium.com
  Subject: Re: [Asterisk-Users] FAX over PRI
 
  On Tuesday 21 March 2006 15:09, Michael Gaudette wrote:
   How should I consider Fax over PRI channels with Asterisk?  Is the
   quality and reliability good, or should I be prepared for alot of
  grief?
 
  I'm having good success doing fax over PRI using a TE405; one span to
  the
  PRI, the other to an FXS channel bank that is almost obscenely
  underutilized
  (3 channels).
 
  I also have channel bank - T100P - IAX2 - TE405 - PRI, where the IAX2
  link
  is a 1-hop SDSL (VOIP only) data link.  This works well too.
 
  -A.
  ___
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 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
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  To UNSUBSCRIBE or update options visit:
 
 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
  The information contained in this email is intended for the personal and
 confidential use
  of the addressee only. It may also be privileged information. If you are
 not the intended
  recipient then you are hereby notified that you have received this
 document in error and
  that any review, distribution or copying of this document is strictly
 prohibited. If you have
  received  this communication in error, please notify Brendata
immediately
 on:
 
  +44 (0)1268 466100, or email '[EMAIL PROTECTED]'
 
  Brendata (UK) Ltd
  Nevendon Hall, Nevendon Road,

RE: [Asterisk-Users] FAX over PRI

2006-05-19 Thread Armin Schindler

On Fri, 19 May 2006, Alexander Lopez wrote:
 I have the same problem, Switched to HylaFax and IAXModem and had MUCH
 better luck. MUCH better being defined as not a single usable fax to
 only missing about 1%. Not bad.
 
Just out of curiousity, if the faxing is not that reliable with softfax 
solutions, why not using hardware DSPs?
E.g. the DIVA Server PRI card (with DSPs for some or all channels) does 
provide faxing without using the CPU (same like the DIVA Server BRI cards).

Armin
  
 
 Spandsp or rather RXFax works on a few machines I hae quite well but on
 others it does not work at all, I have had good retruns with the
 HylaFax.IAXmodem combo on machines that could not use the RxFax. 
 
  
 
  
 
 
 
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Tom
 Christensen
 Sent: Friday, May 19, 2006 12:18 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] FAX over PRI
 
  
 
 I have had nothing but problems receiving faxes over PRIs with spandsp.
 I currently have 4 systems, 4 PRIs from 4 different providers... none of
 them get better than 50% success rates receiving faxes in spandsp, I
 constantly get cut off pages.  No body seems to have a fix for it, and
 it is really frustrating.  Supposedly it is caused by frame slips on
 the PRI, but if that is the case, I am 4 for 4 getting crappy PRIs that
 can't keep time. 
 
 These same boxes work fine when receiving faxes over fxo ports, or if I
 plug a fax machine into an fxs port and call in to a spandsp extension
 the fax will be received just fine, so I am left thinking it must be the
 PRIs, but if all PRIs are this bad, how can anybody be using them? 
 
 Tom
 
 On 5/19/06, Steve Hanselman [EMAIL PROTECTED] wrote:
 
 Sorry for the late reply but both of these are fine, we use spandsp to
 print some faxes and email others.
 
 We also route via a PRI to our other phone system to hylafax on an
 analog modem and also to an analog fax. 
 
 So what you want to do is fine and will work.
 
 Steve
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto: [EMAIL PROTECTED] On Behalf Of Michael
 Gaudette
 Sent: 21 March 2006 20:34
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
 Subject: RE: [Asterisk-Users] FAX over PRI
 
 Hmmm, Im not so sure I can apply this to me though.  I just want to do
 Fax-To-Email using PRI channels as the incoming lines.  Not so much
 transfer
 to a real fax. 
 
 I am assuming that this is easily done with Asterisk? (I did it before
 with
 Asterisk SIP, but it only worked once every 10 tries or so)
 
 Mike
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Andrew
 Kohlsmith
 Sent: March 21, 2006 3:25 PM 
 To: asterisk-users@lists.digium.com
 Subject: Re: [Asterisk-Users] FAX over PRI
 
 On Tuesday 21 March 2006 15:09, Michael Gaudette wrote:
  How should I consider Fax over PRI channels with Asterisk?  Is the 
  quality and reliability good, or should I be prepared for alot of
 grief?
 
 I'm having good success doing fax over PRI using a TE405; one span to
 the
 PRI, the other to an FXS channel bank that is almost obscenely 
 underutilized
 (3 channels).
 
 I also have channel bank - T100P - IAX2 - TE405 - PRI, where the IAX2
 link
 is a 1-hop SDSL (VOIP only) data link.  This works well too.
 
 -A.
 ___ 
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 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:
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 The information contained in this email is intended for the personal and
 confidential use 
 of the addressee only. It may also be privileged information. If you are
 not the intended
 recipient then you are hereby notified that you have received this
 document in error and
 that any review, distribution or copying of this document is strictly
 prohibited. If you have 
 received  this communication in error, please notify Brendata
 immediately on:
 
 +44 (0)1268 466100, or email '[EMAIL PROTECTED]'
 
 Brendata (UK) Ltd
 Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX  UK 
 Registered Office as above. Registered in England No. 2764339
 
 See our current vacancies at www.brendata.co.uk
 ___
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Re: [Asterisk-Users] FAX over PRI

I have a setup like yours as well, IE, I have a fax machine connected into the FXS port, and I can fax and receive faxes to the actual machine 100%it is just receiving faxes in spandsp that exhibits this problem. I don't want to blame spandsp though, because if I send a fax directly from that fax machine to a spandsp extension (from fxs port into asterisk, not traversing the PRI) I get 100% success rates in spandsp. Also, with these machines plugged directly into an FXO analog line, I get 100% success rates in spandsp, it is only over PRIs that I see this problem.
On 5/19/06, Noah Miller [EMAIL PROTECTED] wrote:
Hi Tom - I have had nothing but problems receiving faxes over PRIs with spandsp.I currently have 4 systems, 4 PRIs from 4 different providers... none of them get better than 50% success rates receiving faxes in spandsp, I constantly
 get cut off pages.No body seems to have a fix for it, and it is really frustrating.Supposedly it is caused by frame slips on the PRI, but if that is the case, I am 4 for 4 getting crappy PRIs that can't keep time.
Just an aside thought (sorry to hijack the thread, Steve):50% - Ouch.I only have one PRI at one of our offices, but we use itto receive faxes that are directly sent via Digium FXS to an analogfax machine.I've never formally tallied up the transmission errors,
but we get something close to 100%.Maybe spandsp is an issue here.- NoahOn 5/19/06, Tom Christensen [EMAIL PROTECTED] wrote: I have had nothing but problems receiving faxes over PRIs with spandsp.I
 currently have 4 systems, 4 PRIs from 4 different providers... none of them get better than 50% success rates receiving faxes in spandsp, I constantly get cut off pages.No body seems to have a fix for it, and it is really
 frustrating.Supposedly it is caused by frame slips on the PRI, but if that is the case, I am 4 for 4 getting crappy PRIs that can't keep time. These same boxes work fine when receiving faxes over fxo ports, or if I plug
 a fax machine into an fxs port and call in to a spandsp extension the fax will be received just fine, so I am left thinking it must be the PRIs, but if all PRIs are this bad, how can anybody be using them?
 Tom On 5/19/06, Steve Hanselman [EMAIL PROTECTED] wrote:  Sorry for the late reply but both of these are fine, we use spandsp to
  print some faxes and email others.   We also route via a PRI to our other phone system to hylafax on an  analog modem and also to an analog fax.   So what you want to do is fine and will work.
   Steve-Original Message-  From: [EMAIL PROTECTED]
  [mailto: [EMAIL PROTECTED]] On Behalf Of Michael  Gaudette  Sent: 21 March 2006 20:34  To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
  Subject: RE: [Asterisk-Users] FAX over PRI   Hmmm, Im not so sure I can apply this to me though.I just want to do  Fax-To-Email using PRI channels as the incoming lines.Not so much
  transfer  to a real fax.   I am assuming that this is easily done with Asterisk? (I did it before  with  Asterisk SIP, but it only worked once every 10 tries or so)
   Mike   -Original Message-  From: [EMAIL PROTECTED]  [mailto:
[EMAIL PROTECTED]] On Behalf Of Andrew  Kohlsmith  Sent: March 21, 2006 3:25 PM  To: 
asterisk-users@lists.digium.com  Subject: Re: [Asterisk-Users] FAX over PRI   On Tuesday 21 March 2006 15:09, Michael Gaudette wrote:   How should I consider Fax over PRI channels with Asterisk?Is the
   quality and reliability good, or should I be prepared for alot of  grief?   I'm having good success doing fax over PRI using a TE405; one span to  the
  PRI, the other to an FXS channel bank that is almost obscenely  underutilized  (3 channels).   I also have channel bank - T100P - IAX2 - TE405 - PRI, where the IAX2
  link  is a 1-hop SDSL (VOIP only) data link.This works well too.   -A.  ___  --Bandwidth and Colocation provided by 
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The information contained in this email is intended for the personal and confidential use  of the addressee only. It may also be privileged information. If you are
 not the intended  recipient then you are hereby notified that you have received this document in error and  that any review, distribution or copying of this document is strictly
 prohibited. If you have  receivedthis communication in error, please notify Brendata immediately on:   +44 (0)1268 466100, or email '
[EMAIL PROTECTED]'   Brendata (UK) Ltd  Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BXUK  Registered Office as above. Registered in England No. 2764339
   See our current vacancies at www.brendata.co.uk

Re: [Asterisk-Users] FAX over PRI

Steve,How do I check the counters on clock slips and or CRC errors on the PRI? I'm using digium te110p's for these PRIsTomOn 5/19/06, Steve Hanselman
 [EMAIL PROTECTED] wrote:














We get the occasional bad fax, but it
really is an occasional one, other than that, it's fine.



We don't get any CRC errors or clock
slips on the PRI, I'd certainly say that it would be a good starting
point to check the counters on these, I'd also check that your drives are
using DMA depending on your hardware, we had a customer a while ago who ended
up doing a self install and none of his drives were enabled for DMA.



Steve













From:

[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of 
Tom Christensen
Sent: 19 May 2006 17:18
To: Asterisk Users Mailing List -
 Non-Commercial Discussion
Subject: Re: [Asterisk-Users] FAX
over PRI





I have had nothing but
problems receiving faxes over PRIs with spandsp. I currently have 4
systems, 4 PRIs from 4 different providers... none of them get better than 50%
success rates receiving faxes in spandsp, I constantly get cut off pages.
No body seems to have a fix for it, and it is really frustrating.
Supposedly it is caused by frame slips on the PRI, but if that is
the case, I am 4 for 4 getting crappy PRIs that can't keep time. 

These same boxes work fine when receiving faxes over fxo ports, or if I plug a
fax machine into an fxs port and call in to a spandsp extension the fax will be
received just fine, so I am left thinking it must be the PRIs, but if all PRIs
are this bad, how can anybody be using them? 

Tom



On 5/19/06, Steve
 Hanselman [EMAIL PROTECTED]
wrote:

Sorry for the late reply but both of these are fine, we use spandsp to
print some faxes and email others.

We also route via a PRI to our other phone system to hylafax on an
analog modem and also to an analog fax. 

So what you want to do is fine and will work.

Steve


-Original Message-
From: [EMAIL PROTECTED]
[mailto: [EMAIL PROTECTED]]
On Behalf Of Michael
Gaudette
Sent: 21 March 2006 20:34
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
Subject: RE: [Asterisk-Users] FAX over PRI

Hmmm, Im not so sure I can apply this to me though.I just want to
do
Fax-To-Email using PRI channels as the incoming lines.Not so much
transfer
to a real fax. 

I am assuming that this is easily done with Asterisk? (I did it before
with
Asterisk SIP, but it only worked once every 10 tries or so)

Mike

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]
On Behalf Of Andrew
Kohlsmith
Sent: March 21, 2006 3:25 PM 
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] FAX over PRI

On Tuesday 21 March 2006 15:09, Michael Gaudette wrote:
 How should I consider Fax over PRI channels with Asterisk?Is
the 
 quality and reliability good, or should I be prepared for alot of
grief?

I'm having good success doing fax over PRI using a TE405; one span to
the
PRI, the other to an FXS channel bank that is almost obscenely 
underutilized
(3 channels).

I also have channel bank - T100P - IAX2 - TE405 - PRI, where the IAX2
link
is a 1-hop SDSL (VOIP only) data link.This works well too.

-A.
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[Asterisk-Users] PRI dialing IVR with inband DTMF

2006-05-19 Thread Anthony Cennami

I have a client who is using a Shoretel PBX. This PBX apparently does not send DTMF information OOB, but instead sends this inband via the B-channel.This is traversing an Asterisk box via a PRI. The user calls the IVR (1-800-CALL-DHL), receives audio, but is not able to present DTMF to engage the IVR. With some light research it appears that the DSP is not activating until the call is answered.
DTMF on the SIP side is set to RFC2833 -- calls all work fine when originating from a SIP phone connected to the same device.Any suggestions on what needs to be done to pre-emptively enable DSP and or early media on the PRI (outbound)??
Thanks,Anthony---SIPAsteriskPRIShoretel-- Anthony D Cennami
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[Asterisk-Users] Setup up Intellitouch ITC-3002 Sip phones with Asterisk

2006-05-19 Thread T.S

Sorry if this hit the list twice.. but I didn't see it come through:


Hey guys,
Just for archival purposes, I have setup the Intellitouch ITC-3002 (2006)
SIP phones for use with asterisk (1.2.7.1). After a few gotcha's, I was
able to do transfer's, moh's, push a button to check voicemail, callerID,
etc.. 
One big gotacha was the dial timeout (by default!) is set to 2ms (20s)
and its NOT configurable from the internal web gui on the phone. You have to
download the config, edit it, then up the config again.. 

The line: dial_timeout=your delay in ms will set the dial delay

Another gotcha is sip registration. Even though you set the authentication
ID and password to whatever matches in the sip configured context for this
phone, you still need to put in same username under User name for URL (in
the gui) so it can authenticate. Otherwise it will just run on and on about
can't authenticate, username/auth name mismatch or something.

I hope this helps someone, I couldn't find ANY information on configuring
these phones with asterisk. Plain looking Business 2 line IP phones, but I
couldn't beat the price ($56!) versus the $169 sticker price I have seen
elsewhere.


Terrelle

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Re: [Asterisk-Users] PRI dialing IVR with inband DTMF

2006-05-19 Thread Doug


At 12:06 5/19/2006, Anthony Cennami wrote:
I have a client who is using a Shoretel PBX.  This PBX apparently 
does not send DTMF information OOB, but instead sends this inband 
via the B-channel.


This is traversing an Asterisk box via a PRI.  The user calls the 
IVR (1-800-CALL-DHL), receives audio, but is not able to present 
DTMF to engage the IVR.  With some light research it appears that 
the DSP is not activating until the call is answered.


DTMF on the SIP side is set to RFC2833 -- calls all work fine when 
originating from a SIP phone connected to the same device.


Any suggestions on what needs to be done to pre-emptively enable DSP 
and or early media on the PRI (outbound)??


Thanks,

Anthony

---SIPAsteriskPRIShoretel


Hey Anthony,

I don't know if this will help you but, we had
a hard time getting touchtones (DTMF) to work until
we set both ends to INFO (sometimes called SIP Info)

RFC2833, Inband, Auto, etc. did not work.

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[Asterisk-Users] Re: Non automated call parking

2006-05-19 Thread Steven

Thanks,

I will check them out.

-- 
-- 
Steven

http://www.glimasoutheast.org



Noah Miller [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Hi Steven -

It's not in the general release yet and it doesn't do everything you
want, but check out the metermaid patch
(http://bugs.digium.com/view.php?id=5779).

If you're interested, I also wrote a patch to do one button parking
(http://bugs.digium.com/view.php?id=6340).  You can use it now, but as
the bugs says, it will not be included in future releases as the
applicationmap part of res_features.c will eventually be fixed to do
this correctly.


- Noah



On 5/19/06, Steven [EMAIL PROTECTED] wrote:


 From discussions with the receptionist staff, this is what we need:

 X number of buttons for parking slots.
 These buttons should be lit when a call is parked there.
 When on a call, just pushing an unlit button will park the call. (The do not
 want to hit hold, transfer, etc.)
 Hitting Transfer and the button would most likely be OK.
 To pick up a call, they can push the parking slot button.
 The parking slot should be a dailable extension, so that everyone else can
 just dial 5401, etc to pick up the call.

 I assume that there is some kind of local channel I can do this with and
 avoid any of the automated parking systems like Park,
 ParkandAnnounce or ValetParking, all of which have their pros and cons.

 Digging through res_features.c it is hard to discern where the call really
 is when parked (what kind of channel it is) so I am not
 sure if it can be reproduced in dialplan.

 Has anyone done manual call parking this way?


 Steven

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RE: [Asterisk-Users] Music on Hold restart at beginning for each call

2006-05-19 Thread Tim Sharp

Chris,
The queues idea is a good one.  I will check it out.
Thanks for all of your suggestions.
Tim

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris
Hastie
Sent: Friday, May 19, 2006 3:48 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Music on Hold restart at beginning for
each call


On Tue, 16 May 2006, Tim Sharp [EMAIL PROTECTED] wrote
Chris,
When I tried background it waited until the message was done before 
dialing, just like playback.  Am I missing something?

Wasn't my suggestion :)

If I've understood what you're trying to I would go one of two ways:

Rather than dial each of the four numbers sequentially, dial them 
simultaneously. This should hopefully speed up your average pick up 
time, but will loose any control over preference for who deals with the 
call.

Or investigate queues. I don't have enough people to make it worth my 
while looking at these, so I've no idea if they're what you need, but 
they sound like they might be.
-- 
Chris Hastie
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Re: [Asterisk-Users] PRI dialing IVR with inband DTMF

2006-05-19 Thread Anthony Cennami

Well the communication between the Asterisk and Shoretel is ISDN PRI. If a station on the Shoretel calls a regular number (company auto-attendant, cellphone voicemail, etc) and that number ANSWERS then there do not appear to be any problems with DTMF.
When a station dials an IVR which does not ANSWER but does Early Media, that stations DTMF is not being received by the PRI. My understanding is that this should typically be handled over the D channel, but in a number of test calls I discovered that all DTMF is being sent Inband from Shoretel over the PRI. 
On 5/19/06, Doug [EMAIL PROTECTED] wrote:
At 12:06 5/19/2006, Anthony Cennami wrote:I have a client who is using a Shoretel PBX.This PBX apparentlydoes not send DTMF information OOB, but instead sends this inbandvia the B-channel.
This is traversing an Asterisk box via a PRI.The user calls theIVR (1-800-CALL-DHL), receives audio, but is not able to presentDTMF to engage the IVR.With some light research it appears that
the DSP is not activating until the call is answered.DTMF on the SIP side is set to RFC2833 -- calls all work fine whenoriginating from a SIP phone connected to the same device.Any suggestions on what needs to be done to pre-emptively enable DSP
and or early media on the PRI (outbound)??Thanks,Anthony---SIPAsteriskPRIShoretelHey Anthony,I don't know if this will help you but, we had
a hard time getting touchtones (DTMF) to work untilwe set both ends to INFO (sometimes called SIP Info)RFC2833, Inband, Auto, etc. did not work.
-- Anthony D Cennami
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Re: [Asterisk-Users] X100P not recognised on FreeBSD system


On Fri, 19 May 2006, Dinesh Nair [EMAIL PROTECTED] wrote


On 05/19/06 18:57 Chris Hastie said the following:

Yes, I have these. The modules load, but ztcfg complains
ZT_CHANCONFIG failed on channel 1: No such device or address (6) and as I
said, it doesn't appear that the card has been recognised by the kernel.


could you try the X100P in anther system to rule out issues with the 
Via board you're using ?




best I can manage is a very old dell optiplex gxi, and it's not 
recognised in that either. Time to assume a dodgy card?

--
Chris Hastie
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RE: [Asterisk-Users] FAX over PRI

For me the answer is simple:

1   Not open source, IT is a thrill to be able to say what you did
without Closed SW.

2   Expensive

Eicon Diva Server Analog- 4 Port Price: $1,695.00 
Most customers won't need that level of Fax anyway, it is
usually a freebee tossed in as Splenda (ie Artificial Sweetener)

3   Not that big of a deal, most people would have an investment in
a fax machine anyway.

Just my .10 (Had to increase 5 fold to pay for gas for SUV)


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Armin Schindler
 Sent: Friday, May 19, 2006 12:54 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] FAX over PRI
 
 On Fri, 19 May 2006, Alexander Lopez wrote:
  I have the same problem, Switched to HylaFax and IAXModem and had
MUCH
  better luck. MUCH better being defined as not a single usable fax to
  only missing about 1%. Not bad.
 
 Just out of curiousity, if the faxing is not that reliable with
softfax
 solutions, why not using hardware DSPs?
 E.g. the DIVA Server PRI card (with DSPs for some or all channels)
does
 provide faxing without using the CPU (same like the DIVA Server BRI
 cards).
 
 Armin
 
 
  Spandsp or rather RXFax works on a few machines I hae quite well but
on
  others it does not work at all, I have had good retruns with the
  HylaFax.IAXmodem combo on machines that could not use the RxFax.
 
 
 
 
 
  
 
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of Tom
  Christensen
  Sent: Friday, May 19, 2006 12:18 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [Asterisk-Users] FAX over PRI
 
 
 
  I have had nothing but problems receiving faxes over PRIs with
spandsp.
  I currently have 4 systems, 4 PRIs from 4 different providers...
none of
  them get better than 50% success rates receiving faxes in spandsp, I
  constantly get cut off pages.  No body seems to have a fix for it,
and
  it is really frustrating.  Supposedly it is caused by frame slips
on
  the PRI, but if that is the case, I am 4 for 4 getting crappy PRIs
that
  can't keep time.
 
  These same boxes work fine when receiving faxes over fxo ports, or
if I
  plug a fax machine into an fxs port and call in to a spandsp
extension
  the fax will be received just fine, so I am left thinking it must be
the
  PRIs, but if all PRIs are this bad, how can anybody be using them?
 
  Tom
 
  On 5/19/06, Steve Hanselman [EMAIL PROTECTED] wrote:
 
  Sorry for the late reply but both of these are fine, we use spandsp
to
  print some faxes and email others.
 
  We also route via a PRI to our other phone system to hylafax on an
  analog modem and also to an analog fax.
 
  So what you want to do is fine and will work.
 
  Steve
 
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto: [EMAIL PROTECTED] On Behalf Of
Michael
  Gaudette
  Sent: 21 March 2006 20:34
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
  Subject: RE: [Asterisk-Users] FAX over PRI
 
  Hmmm, Im not so sure I can apply this to me though.  I just want to
do
  Fax-To-Email using PRI channels as the incoming lines.  Not so much
  transfer
  to a real fax.
 
  I am assuming that this is easily done with Asterisk? (I did it
before
  with
  Asterisk SIP, but it only worked once every 10 tries or so)
 
  Mike
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of Andrew
  Kohlsmith
  Sent: March 21, 2006 3:25 PM
  To: asterisk-users@lists.digium.com
  Subject: Re: [Asterisk-Users] FAX over PRI
 
  On Tuesday 21 March 2006 15:09, Michael Gaudette wrote:
   How should I consider Fax over PRI channels with Asterisk?  Is the
   quality and reliability good, or should I be prepared for alot of
  grief?
 
  I'm having good success doing fax over PRI using a TE405; one span
to
  the
  PRI, the other to an FXS channel bank that is almost obscenely
  underutilized
  (3 channels).
 
  I also have channel bank - T100P - IAX2 - TE405 - PRI, where the
IAX2
  link
  is a 1-hop SDSL (VOIP only) data link.  This works well too.
 
  -A.
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and
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are
  not the intended
  recipient then you are hereby notified that you have received this
  document in error and
  that any

RE: [Asterisk-Users] PRI dialing IVR with inband DTMF









Is it Phone - ShoreTel - Asterisk
- PSTN ???















From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony Cennami
Sent: Friday, May 19, 2006 1:47 PM
To: Doug
Cc: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] PRI
dialing IVR with inband DTMF





Well the communication
between the Asterisk and Shoretel is ISDN PRI. If a station on the
Shoretel calls a regular number (company auto-attendant, cellphone voicemail,
etc) and that number ANSWERS then there do not appear to be any
problems with DTMF. 

When a station dials an IVR which does not ANSWER but does Early
Media, that stations DTMF is not being received by the PRI. My
understanding is that this should typically be handled over the D channel, but
in a number of test calls I discovered that all DTMF is being sent Inband from
Shoretel over the PRI. 






On 5/19/06, Doug
[EMAIL PROTECTED] wrote:

At 12:06 5/19/2006,
Anthony Cennami wrote:
I have a client who is using a Shoretel PBX.This PBX apparently
does not send DTMF information OOB, but instead sends this inband
via the B-channel.
 
This is traversing an Asterisk box via a PRI.The user calls the
IVR (1-800-CALL-DHL), receives audio, but is not able to present
DTMF to engage the IVR.With some light research it appears that
the DSP is not activating until the call is answered.

DTMF on the SIP side is set to RFC2833 -- calls all work fine when
originating from a SIP phone connected to the same device.

Any suggestions on what needs to be done to pre-emptively enable DSP 
and or early media on the PRI (outbound)??

Thanks,

Anthony

---SIPAsteriskPRIShoretel

Hey Anthony,

I don't know if this will help you but, we had 
a hard time getting touchtones (DTMF) to work until
we set both ends to INFO (sometimes called SIP Info)

RFC2833, Inband, Auto, etc. did not work.






-- 
Anthony D Cennami








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[Asterisk-Users] Re: Non automated call parking

2006-05-19 Thread Steven

I'll have to test this over the weekend since it affects so many files.

If I knew C programming better or the guts of asterisk better, I would attempt 
to write this myself.

The root issue is that all various parking techniques are based off of the 
parking in features.conf.
What if this parking style is not preferred?

I assume that it would be easier to not use the standard park logic and to just 
do the following.

Make extensions that can hold a call. (like a 701)
Make this extension hintable for use in button programming.
If I am on a call and hit a non-lit button, it parks it there.
If I am not on a call and push the lit button, I connect to the park.
I suppose that if you are on a call and hit a lit button, it should either not 
be processed, or should join as a three way call. 
(either logic is justifiable)
These park extensions should still be callable so analog, etc. extensions can 
also connect to them.


-- 
-- 
Steven

http://www.glimasoutheast.org



Noah Miller [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Hi Steven -

It's not in the general release yet and it doesn't do everything you
want, but check out the metermaid patch
(http://bugs.digium.com/view.php?id=5779).

If you're interested, I also wrote a patch to do one button parking
(http://bugs.digium.com/view.php?id=6340).  You can use it now, but as
the bugs says, it will not be included in future releases as the
applicationmap part of res_features.c will eventually be fixed to do
this correctly.


- Noah



On 5/19/06, Steven [EMAIL PROTECTED] wrote:


 From discussions with the receptionist staff, this is what we need:

 X number of buttons for parking slots.
 These buttons should be lit when a call is parked there.
 When on a call, just pushing an unlit button will park the call. (The do not
 want to hit hold, transfer, etc.)
 Hitting Transfer and the button would most likely be OK.
 To pick up a call, they can push the parking slot button.
 The parking slot should be a dailable extension, so that everyone else can
 just dial 5401, etc to pick up the call.

 I assume that there is some kind of local channel I can do this with and
 avoid any of the automated parking systems like Park,
 ParkandAnnounce or ValetParking, all of which have their pros and cons.

 Digging through res_features.c it is hard to discern where the call really
 is when parked (what kind of channel it is) so I am not
 sure if it can be reproduced in dialplan.

 Has anyone done manual call parking this way?


 Steven

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Re: [Asterisk-Users] PRI dialing IVR with inband DTMF

2006-05-19 Thread Anthony Cennami

Two situations:Shoretel Phone -- ShoreTel -- PRI -- Asterisk -- Softswitch -- VoIP/PSTN -- This situation does NOT work. User hears audio, but Asterisk does not appear to process DTMF. Note, this is ONLY on IVR applications where we are not getting 200/connect passed through even though we're hearing IVR audio (early media)
Second situation:IP Phone -- Asterisk -- Softswitch -- VoIP/PSTN -- Works as expected. You can dial the IVR and still send DTMF.It seems like there are some early media problems when routing traffic through the PRI.
On 5/19/06, Alexander Lopez [EMAIL PROTECTED] wrote:















Is it Phone - ShoreTel - Asterisk
- PSTN ???















From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of 
Anthony Cennami
Sent: Friday, May 19, 2006 1:47 PM
To: Doug
Cc: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] PRI
dialing IVR with inband DTMF





Well the communication
between the Asterisk and Shoretel is ISDN PRI. If a station on the
Shoretel calls a regular number (company auto-attendant, cellphone voicemail,
etc) and that number ANSWERS then there do not appear to be any
problems with DTMF. 

When a station dials an IVR which does not ANSWER but does Early
Media, that stations DTMF is not being received by the PRI. My
understanding is that this should typically be handled over the D channel, but
in a number of test calls I discovered that all DTMF is being sent Inband from
Shoretel over the PRI. 






On 5/19/06, Doug
[EMAIL PROTECTED] wrote:

At 12:06 5/19/2006,
Anthony Cennami wrote:
I have a client who is using a Shoretel PBX.This PBX apparently
does not send DTMF information OOB, but instead sends this inband
via the B-channel.
 
This is traversing an Asterisk box via a PRI.The user calls the
IVR (1-800-CALL-DHL), receives audio, but is not able to present
DTMF to engage the IVR.With some light research it appears that
the DSP is not activating until the call is answered.

DTMF on the SIP side is set to RFC2833 -- calls all work fine when
originating from a SIP phone connected to the same device.

Any suggestions on what needs to be done to pre-emptively enable DSP 
and or early media on the PRI (outbound)??

Thanks,

Anthony

---SIPAsteriskPRIShoretel

Hey Anthony,

I don't know if this will help you but, we had 
a hard time getting touchtones (DTMF) to work until
we set both ends to INFO (sometimes called SIP Info)

RFC2833, Inband, Auto, etc. did not work.






-- 
Anthony D Cennami









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Re: [Asterisk-Users] Re: Non automated call parking

2006-05-19 Thread Dr. Michael J. Chudobiak


Steven wrote:

Make extensions that can hold a call. (like a 701)
Make this extension hintable for use in button programming.
If I am on a call and hit a non-lit button, it parks it there.
If I am not on a call and push the lit button, I connect to the park.
I suppose that if you are on a call and hit a lit button, it should either not be processed, or should join as a three way call. 
(either logic is justifiable)

These park extensions should still be callable so analog, etc. extensions can 
also connect to them.


Steven,

That arrangement would be great, but right now the closest existing 
method is the metermaid patch at 
http://bugs.digium.com/view.php?id=5779, and it looks like that won't 
even make it into 1.4. Sigh. (oej: people need this patch!)


I created a bounty two years ago at 
http://www.voip-info.org/wiki/view/Asterisk+bounty+snom+call+park for an 
arrangement like you describe, but there was no interest, so I dropped 
sponsorship of it. You can take over the bounty if you like.


I'm using the metermaid patch on 1.2.6, and it works very nicely with my 
Snom 360s. (Press a 700 button to park, and observe the 701-7xx button 
LEDs to see parking slot status).


- Mike
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Re: [Asterisk-Users] FAX over PRI

If this works reliably while rxfax+spandsp does not, wouldn't this point the blame at rxfax as opposed to spandsp?IAXmodem uses spandsp the same way rxfax does right?On 5/19/06, 
Alexander Lopez [EMAIL PROTECTED] wrote:



















I have the same problem, Switched to
HylaFax and IAXModem and had MUCH better luck. MUCH better being defined as not
a single usable fax to only missing about 1%. Not bad.



Spandsp or rather RXFax works on a few
machines I hae quite well but on others it does not work at all, I have had
good retruns with the HylaFax.IAXmodem combo on machines that could not use the
RxFax. 















From:
[EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED]]
On Behalf Of Tom Christensen
Sent: Friday, May 19, 2006 12:18
PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] FAX
over PRI





I have had nothing but
problems receiving faxes over PRIs with spandsp. I currently have 4
systems, 4 PRIs from 4 different providers... none of them get better than 50%
success rates receiving faxes in spandsp, I constantly get cut off pages.
No body seems to have a fix for it, and it is really frustrating. Supposedly
it is caused by frame slips on the PRI, but if that is the case, I
am 4 for 4 getting crappy PRIs that can't keep time. 

These same boxes work fine when receiving faxes over fxo ports, or if I plug a
fax machine into an fxs port and call in to a spandsp extension the fax will be
received just fine, so I am left thinking it must be the PRIs, but if all PRIs
are this bad, how can anybody be using them? 

Tom



On 5/19/06, Steve
Hanselman [EMAIL PROTECTED]
wrote:

Sorry for the late reply but both of these are fine, we use spandsp to
print some faxes and email others.

We also route via a PRI to our other phone system to hylafax on an
analog modem and also to an analog fax. 

So what you want to do is fine and will work.

Steve


-Original Message-
From: [EMAIL PROTECTED]
[mailto: [EMAIL PROTECTED]]
On Behalf Of Michael
Gaudette
Sent: 21 March 2006 20:34
To: 'Asterisk Users Mailing List - Non-Commercial
 Discussion' 
Subject: RE: [Asterisk-Users] FAX over PRI

Hmmm, Im not so sure I can apply this to me though.I just want to
do
Fax-To-Email using PRI channels as the incoming lines.Not so much
transfer
to a real fax. 

I am assuming that this is easily done with Asterisk? (I did it before
with
Asterisk SIP, but it only worked once every 10 tries or so)

Mike

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]
On Behalf Of Andrew
Kohlsmith
Sent: March 21, 2006 3:25 PM 
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] FAX over PRI

On Tuesday 21 March 2006 15:09, Michael Gaudette wrote:
 How should I consider Fax over PRI channels with Asterisk?Is
the 
 quality and reliability good, or should I be prepared for alot of
grief?

I'm having good success doing fax over PRI using a TE405; one span to
the
PRI, the other to an FXS channel bank that is almost obscenely 
underutilized
(3 channels).

I also have channel bank - T100P - IAX2 - TE405 - PRI, where the IAX2
link
is a 1-hop SDSL (VOIP only) data link.This works well too.

-A.
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Re: [Asterisk-Users] Experience with IBM X346 machines and Sangoma

2006-05-19 Thread Rich Adamson


Cosmin Prund wrote:
I wanted to see where those periodical spikes are coming from so I 
started shutting things down. The first thing to go was Asterisk. Next 
went many different things like mysql and ntpd. Finally I killed zaptel 
(/etc/init.d/zaptel stop) - and the spiking stoped!


Next I rebooted and I've done /etc/init.d/zaptel stop straight away. 
The spiking stoped again. I've done /etc/init.d/zaptel start and 
spiking started again!


Is there something funny happening with my zaptel?
Wolfgang Zweimueller, can you give this a try too? Does your spiking 
stop when you stop zaptel?


There have been multiple threads over the last two years about the exact 
same 'vmstat 1' results, and no one has ever come up with a logical 
explanation as to why it occurs.


Of the several (probably hundreds) of posts in the past, it does not 
seem to be a linux distro issue, and stopping zaptel always removes the 
symptom.


It seems the majority of folks that were involved with this in the past 
'assumed' the results were what was impacting fax through the TDM400. 
But, don't think anyone proved that.


No other guesses at this time.

R.

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RE: [Asterisk-Users] RTP Packetization

2006-05-19 Thread Dan Austin

Sorry for the top post, i've only got a few seconds to respond.

The Async patch (-I) has nothing to do with packetization.  The
poster that added that information to the bug notes under 5162
was confused.

As to why it is not working, you said you set it on a peer.
Did that 'peer' call Asterisk, or did another device on
Asterisk call it?

Is the second device also using 30ms?  Do you have re-invites
enabled?  A re-invite to/from a device not told to use 30ms
won't use 30ms.

I use type 'friend' and get 30ms to/from my endpoints, and
since my server is primarily for MeetMe, I do not have reinvite
enabled.

Dan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Patrick
Neubauer
Sent: Friday, May 19, 2006 7:27 AM
To: Asterisk Users Mailing List
Subject: [Asterisk-Users] RTP Packetization

Hi all,

I need to be able to adjust packet sizes and found the patch at 
http://bugs.digium.com/view.php?id=5162

Thus, I checked out and compiled 
http://svn.digium.com/view/asterisk-old/team/group/5162_rtp_packetizatio
n

I added the line packetization = 30 for one peer in my sip.conf and 
started asterisk with the -I switch for async RTP.
That's all it takes according to the 5162 issue page. Nevertheless, 
asterisk still keeps sending it 20ms packets, even though a sip show 
peer foobar shows Packetization: 30.

What could be wrong? What about that ztdummy thing for internal timing? 
Is this necessary to run asterisk properly? Is it important for 
packetization?

Regards, Patrick
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RE: [Asterisk-Users] FAX over PRI