[asterisk-users] Is this phone any good

[Tim]

Hopefully this didn't get posted to the list already. I think I was having some
email problems and lost most of the days postings.

Anyway, I was wondering is anyone knew if the Gnet VP320S phones are any good?

Tim
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[asterisk-users] pstn failback

[stan ford]

On fonalities web page, i see they offer pstn failback as a feature of their asterisk package. i've also heard before of failing back to a pri line if your t1 voip line fails. my question is. in order to have pstn or pri failback, dont you basically have to have all the equipment there on standby, a PRI line, TDM cards, PRI/T1 cards, a bunch of digital or analong phones. it just seems like a whole lot of hardware to be sitting there waiting for a disaster. unless im just not understanding pstn/pri failback. can someone shed some light?     also, if you've got a dedicated full t1 line for voice, and have a low amount of users for that t1, is there really to worry about failing back to a pstn? seeing how reliable a t1 is. is anyone out here using full voip telelphony solution only?     thanks. 
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Re: [asterisk-users] asterisk on 2.4 kernel ... scheduler problem?

[Cesc]

inline ...

On 9/28/06, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:

On Thu, Sep 28, 2006 at 03:24:27PM +0200, Cesc wrote:
> Hello people!
>
> I have an inquiry (not a doubt ;D ). Actually, two.
>
> I am trying to run asterisk on an embedded Power PC platform on which
> we have a linux with a 2.4.2x kernel.

Still uses 2.4 today? Not a very good sign.



yeah, i know ... but multiple issues here: management and money;
backwards compatibility (not all sectors are concerned about cutting
edge) and reduced size of the 2.4 kernel. Anyway ... it is just the
way it is :)


> In there, the linux scheduler
> runs at 100Hz. On a 2.6 kernel, the scheduler is at 1KHz. I just take
> this from a colleague ... hope it is true :)  I only need to run the
> VOIP part, thus no POTS or external hardware. Actually, I just need
> SIP and H323 (channels/h323). Is there any problem to be expected from
> the scheduler difference? Or any other from running on a 2.4 kernel?
> Some colleague said that asterisk needs the 1KHz scheduler, but i
> cannot believe that it won't run on a 2.4 kernel ... Anyway, that is
> why i am asking.

If you really want a 1kHz timing source for 2.4, build zaptel. But
you'll need a USB UHCI chip.


my question is, do i really need the 1Khz scheduler? remember i just
want to operate SIP and H323 ...
If i do really need it ... there is NO chance to add any extra
software or the like. It has to be all software based, otherwise it is
a nono.

No one else uses asterisk with a 2.4 kernel? forget the Power PC story
.. but just 2.4?


>
> The other inquiry ... as the system is embedded we have not so much
> disk space available. So, i need a minimal asterisk installation. When
> compiled and stripped, the biggest amount of space is taken by the
> modules. My question is, can asterisk work with just the chan_sip.so,
> chan_h323.so and the codec_*.so? is there any other module needed? I
> need only be able to bridge sip to h323, no extra fancy stuff needed
> (parking, echo, blah, blah, ... )

Don't autoload modules in modules.conf . Load only the modules you need.
Use 'load' from the CLI to manually load modules to see if you need
them.

One shortcut you may take is to use DeStar. It is an Asterisk
configuration generator that generates a configuration with explicit
"load" in modules.conf, rather than loading everything...


ok, i will do that ... i will try to cut to a bare minimum. Tks!

REgards,

Cesc
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Re: [asterisk-users] Building the Perfect Box

[Conrad Wood]

> > 1. Good box, see above

We used IBM, HP/Compaq and Fujitsu Siemens. None of them came close to
supermicro opteron servers. The Serverworks-HT1000 Chipset rocks (apart
from the broadcom nic). Things "just work" and I tell it exactly which
IRQs to use for which slot. 
And boy, do they feel fast when working with. Actual spec isn't that
impressive but the whole board is designed so well it easily outperforms
any HP, Fujitsu,Dell or IBM I've seen so far.


> > 2. Good LAN - this is so critical and so often overlooked in the day and age
> > of guys crimping their own cables and running $150 switches. You can't do
> > that, and if you do, you do so at your own peril. Managed swiches,
> > professional cable installation. This is not a problem for me since I *am* a
> > professional cable installer but I have actually witnessed people making
> > patch cables with a flat blade screwdriver and a hammer!

Did we meet? ;-) I used to do that in the 90s with coax wiring until I
finally saw the light ;)
And then I wondered why my fluke tester said "no".
Seriously though, you're quite right decent cabling is _essential_

> 
> > 3. Tuning of the LAN - VLAN's are good. QoS packets are good. Switches that
> > honor the QoS packets are good. 
> 
I tend to use a different switch altogether and lock the switch ports
(because people do plug weird stuff in which suddenly acts as a dhcp
server or does other annoying things)

> 
> > 4. Handset selection - this is another biggie. I've selected Snom 360's, and
> > yes they have warts, but they are feature rich for the price and Snom is

snom 360s are definitely the best, but our clients seem to prefer the
look and feel of cisco 79xx. 

> > USERS INVOLVED.
> > 5. Tuning of Asterisk box itself - this cannot be under emphasized. This is
> > a very important step and tuning methodologies vary according to distro,
> > skill of the admin, and particular circumstances. I've learned *way* more
> > than I ever wanted to about processor affinity sinc I started using
> > Asterisk. 
> 
I install the box minimalistic to begin with (debian usually). Compiling
a preemptive kernel helps too. Stop unnecessary daemons and off it goes.

> 
> > 6. Termination of PSTN. Basically I would never do an Asterisk install where
> > I was forced to do something stupid like aggregate a dozen Centrex lines or
> > some mickey mouse deal with FXO ATA's or whatever except for a hobby or
> > prototype install. PRI, BRI, IAX or SIP, don't mess around with anything
> > else. 
> 
absolutely. 

> > 7. Relationship with provider. What is their SLA? Is it the incumbent or the
> > clec? An incumbent will be more expensive and more difficult to deal with
> > but they will tend to be more reliable. A clec will be cheaper and they will
> > be way more accomodating but you will most likely not get five 9's from
> > them. A VoIP provider should never be trusted, period. You will not get five
> > nines from them, ever. Plan failover situations accordingly. 
> 
> "SLA?  What's a 'SLA'?"  :-)

Service Level Agreement. Normally it means if your line fails you get
£50 or so. Maybe £100. But *never* enough to compensate for the trouble.
You have to have a backup plan.

> 
> Amusingly, a client's * box went down this morning.  I didn't get the
> washout, but the mitigation wasn't well planned either -- everyone with
> an Asterisk box should know what they're going to do if it falls over,
> in detail.  In a notebook.  Just like when the nuclear missles start
> going.

Yes of course. These notebooks tend to get forgotten in a cupboard until
the day they're needed. And then they're so out of date that they're
more damaging that useful.
Do you update the text on the server itself somewhere? How do you go
about keeping it up-to-date? "Just discipline"? Do you work in a team
with others?

> 
> > 8. Plan plan plan plan. A good install of ANYTHING is 80% planning 20% doing
> > it. What is your plan when your primary PSTN provider fails? What is your
> > plan if your Asterisk box goes pear shaped? My dialplan can survive either
> > PSTN, WAN or LAN failure (albeit with reduced functionality). I also keep a
> > cold spare, an identically configured box that I can literally throw into
> > the rack, turn it on, plug in the PRI's and no problem. 

Planning is vital, but do you play through disaster recovery scenarios
regularly ?

> I was musing on giving station users a list of pseudo-CLASS dialcodes
> they could punch to mark that there was a problem with a previous call,
> so it would go into the logs and could be checked latter.

This is actually quite a cool idea.  
That's almost a MOS test ;) Press 1-5 to rate the call ;-)))
Would be even better if I could do that *during* the call already. Maybe
in features.conf...hm.

Conrad



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Re: [asterisk-users] WAS: 64 analog phones NOW: Selection criteri a and recipie for a good Asterisk install [long]

[Marnus van Niekerk]

Colin,

for the record I think this post was exellent and deserves a
compliment.  It is probably one of the best outlines of what is needed
for a professional system I have ever seen.

Marnus van Niekerk

Colin Anderson wrote:

  
I concur with your approach, but "Tier 1" means as little here as it
does when evaluating Internet backbone carriers.  could you expand on
what evaluation criteria you use?  I'm going to be pre-speccing some
stuff myself this month...

  
  
Sorry I should have been more clear. A good Asterisk install needs a
holistic approach to use a hippy dippy phrase. A Tier 1 server, which is a
midrange to high end name brand server from the Big 3 (Dell, HP/Compaq, IBM,
am I missing someone?) is usually highly optimized for bus bandwidth
although that design was intended for a different use - usually massive disk
I/O. As well, a Tier 1 server will have two seperate, independent PCI buses
and this to me is a critical feature - it allows you to completely separate
your TDM traffic from network, disk I/O etc. On my big production Netfinity,
I took great care to ensure the Digium cards were all on their lonesome on a
single bus, and everything else on the other bus. This is how I can run two
TE110's in a single box with no problems. zttest does not give me 100% all
the time, but on the other hand it *never* drops below 99.9987%, even under
load. I selected this Netfinity because of the obvious care put into it's
design, but the specs are unimpressive: quad Xeon 700's. CPU is over rated
for Asterisk, IMO unless you are doing tons of transcoding and if you are
doing that, then your design is flawed. 

Anyway, the holistic approach (to go on a small rant for the newbie lurkers)
be summed up as follows:

1. Good box, see above
2. Good LAN - this is so critical and so often overlooked in the day and age
of guys crimping their own cables and running $150 switches. You can't do
that, and if you do, you do so at your own peril. Managed swiches,
professional cable installation. This is not a problem for me since I *am* a
professional cable installer but I have actually witnessed people making
patch cables with a flat blade screwdriver and a hammer!
3. Tuning of the LAN - VLAN's are good. QoS packets are good. Switches that
honor the QoS packets are good. 
4. Handset selection - this is another biggie. I've selected Snom 360's, and
yes they have warts, but they are feature rich for the price and Snom is
really good about revising firmware. When you select handsets, GET YOUR
USERS INVOLVED.
5. Tuning of Asterisk box itself - this cannot be under emphasized. This is
a very important step and tuning methodologies vary according to distro,
skill of the admin, and particular circumstances. I've learned *way* more
than I ever wanted to about processor affinity sinc I started using
Asterisk. 
6. Termination of PSTN. Basically I would never do an Asterisk install where
I was forced to do something stupid like aggregate a dozen Centrex lines or
some mickey mouse deal with FXO ATA's or whatever except for a hobby or
prototype install. PRI, BRI, IAX or SIP, don't mess around with anything
else. 
7. Relationship with provider. What is their SLA? Is it the incumbent or the
clec? An incumbent will be more expensive and more difficult to deal with
but they will tend to be more reliable. A clec will be cheaper and they will
be way more accomodating but you will most likely not get five 9's from
them. A VoIP provider should never be trusted, period. You will not get five
nines from them, ever. Plan failover situations accordingly. 
8. Plan plan plan plan. A good install of ANYTHING is 80% planning 20% doing
it. What is your plan when your primary PSTN provider fails? What is your
plan if your Asterisk box goes pear shaped? My dialplan can survive either
PSTN, WAN or LAN failure (albeit with reduced functionality). I also keep a
cold spare, an identically configured box that I can literally throw into
the rack, turn it on, plug in the PRI's and no problem. 
9. Internet bandwidth and latency. I am fortunate enough to have a great IP
provider. Ask for demos - most guys will install a 90 day trial or something
like that. Do not believe the brochure, get the product installed and put it
under load. 
10. Traffic prioritization at the IP demarc - total no brainer. 
11. Constant, constant user feedback and remediation. If you are not talking
to your users, your install will ultimately fail even if you have the best
of everything. Underpromise and overdeliver. Never loose sight of the basics
- they have to pick up the phone, and it has to work. Always. 
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Re: [asterisk-users] WAS: 64 analog phones NOW: Selection criteri a and recipie for a good Asterisk install [long]

[Raphaël Jacquot]

Colin Anderson wrote:
>> I concur with your approach, but "Tier 1" means as little here as it
>> does when evaluating Internet backbone carriers.  could you expand on
>> what evaluation criteria you use?  I'm going to be pre-speccing some
>> stuff myself this month...
> 
> Sorry I should have been more clear. A good Asterisk install needs a
> holistic approach to use a hippy dippy phrase. A Tier 1 server, which is a
> midrange to high end name brand server from the Big 3 (Dell, HP/Compaq, IBM,
> am I missing someone?) is usually highly optimized for bus bandwidth
> although that design was intended for a different use - usually massive disk
> I/O. As well, a Tier 1 server will have two seperate, independent PCI buses
> and this to me is a critical feature - it allows you to completely separate
> your TDM traffic from network, disk I/O etc. On my big production Netfinity,

nothing a good opteron motherboard from tyan can't do (something like
http://tyan.com/products/html/thunderk8we.html )

> 5. Tuning of Asterisk box itself - this cannot be under emphasized. This is
> a very important step and tuning methodologies vary according to distro,
> skill of the admin, and particular circumstances. I've learned *way* more
> than I ever wanted to about processor affinity sinc I started using
> Asterisk. 

I'll be interested in more pointers on that one

> 6. Termination of PSTN. Basically I would never do an Asterisk install where
> I was forced to do something stupid like aggregate a dozen Centrex lines or
> some mickey mouse deal with FXO ATA's or whatever except for a hobby or
> prototype install. PRI, BRI, IAX or SIP, don't mess around with anything
> else. 

sometimes you don't really have a choice. some providers don't know what
PRI is

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Re: [asterisk-users] pstn failback

[Lacy Moore]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

stan ford wrote:
> On fonalities web page, i see they offer pstn failback as a feature of their 
> asterisk package. i've also heard before of failing back to a pri line if 
> your t1 voip line fails. my question is. in order to have pstn or pri 
> failback, dont you basically have to have all the equipment there on standby, 
> a PRI line, TDM cards, PRI/T1 cards, a bunch of digital or analong phones. it 
> just seems like a whole lot of hardware to be sitting there waiting for a 
> disaster. unless im just not understanding pstn/pri failback. can someone 
> shed some light?
>
>   also, if you've got a dedicated full t1 line for voice, and have a low 
> amount of users for that t1, is there really to worry about failing back to a 
> pstn? seeing how reliable a t1 is. is anyone out here using full voip 
> telelphony solution only?
>
Having had our XO connection go down within a week or so of switching to
a PRI, I can see how having a fallback would be good.  It was down for
about 4 hours.  However, that was back in May or June and hasn't been
down since.  I couldn't justify having something on standby for our
business.  But, our clients can reach us by phone in the office or cell,
and we can easily make outgoing calls on our cellphones. Our type of
business is not really dependent on absolutely having the phones up 100%
of the time.

If you do get a fallback, get it from different providers.  If your T1
from provider X is down, chances are a PRI from provider X will be
following the same path and be down as well.
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[asterisk-users] Sirrix to Legacy PBX

[Marnus van Niekerk]

Hi,

I have a site where we currently have * and a legacy PBX (Samsung DCS)
both with BRI lines coming in.  The two in linked together with four
analogue channels which works most of the time.

The local Telco in South Africa will (eventually) install a PRI with
100DID's next week and we want to then use the existing sirrix board to
feed 4BRI's to the legacy PBX effectively using it a a channel bank for
the existing phones on that system.

Something like this:

PRI --> * --> sirrix 4BRI --> DCS

The DCS will think it is connected to the telco and make calls as usual
(except for calls to internal extensions on * but that is a DCS
programming issue.)


Anybody used a sirrix board like the before and any pointers before we
attempt this?


Thank you


Marnus van Niekerk

-- 

"Opportunity is missed by most people because it is
dressed in overalls and looks like work."

Thomas Alva Edison - Inventor of 1093 patents,
including the light bulb, phonogram and motion pictures.




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[asterisk-users] Difference between "SIP Server" and "Outbound Proxy"

[Brian Candler]

It seems a number of phones have separate settings for "SIP Server" and
"Outbound proxy".

More specifically, the Linux X-Lite softphone I've been playing with has
"SIP Proxy" and "Out Bound Proxy" settings, and it seems the GrandStream
BudgeTone 100 has "SIP Server" and "Outbound Proxy"; see
http://siproxd.sourceforge.net/siproxd_guide/siproxd_guide_c6s2.html

Now, my question is: what exactly is the distinction? Reading RFC 3261, the
concept of an "outbound proxy" is clearly defined - all outbound SIP packets
will have their destination set to this address. I'm fine with that. But
what's the separate "SIP Proxy" or "SIP Server" setting for?

By a process of experimentation on X-Lite, I have determined:

* All outbound requests are sent to the "outbound proxy", unless I have
  set "use outbound proxy: never", in which case they're sent to the
  "SIP proxy"

* I don't see any other circumstance in which the "SIP proxy" setting is
  used. I've done some tcpdumping of SIP traffic, and may have missed
  something. However if I point the "SIP server" setting to some other
  host in my network, I don't see its address being used at all - neither
  registration nor call origination.

More specifically: if I set the "Outbound proxy" to 10.69.255.251 (an
Asterisk server), and the "SIP proxy" to 10.69.255.252 (another PC), then
all REGISTER and INVITE requests are sent to 10.69.255.251. No packets are
sent to 10.69.255.252, and as far as I can tell, none of the packets to
10.69.255.251 include 10.69.255.252 in their contents either.

So I don't understand what the other "SIP Proxy" or "SIP Server" setting
is for, and I couldn't find that info on voip-info.org or google.

If anyone could explain, clearly and precisely, how this setting is used by
the client (in terms of how it affects the SIP exchanges), I'd be very
grateful!

Many thanks,

Brian Candler.
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[asterisk-users] re: asterisk/SER integration - HELP

[Yair Hakak]

hi list,
 i need some help here...
 
i have the following setup
 
1. openser running on port 5060 - succesfully registering endpoints. all good.
2 asterisk 1.2 running sip on port 5070 on the same machine.
3. asterisk 1.09 running sip on port 5070 a different machine.
 
i have 2 routes in my openser.cfg:
1:
 if (uri =~ "sip:[EMAIL PROTECTED]")    {    log(1, "Forwarding to Asterisk\n");  rewritehostport("asterisk1IP:5070");
 route(1);    return;    }
 
2:
  if (uri =~ "sip:[EMAIL PROTECTED]")    {    log(1, "Forwarding to Asterisk\n");  rewritehostport("asterisk2IP:5070");
 route(1); return;    } 
route 2 works fine (to asterisk 1.09).
sip.conf on asterisk 1.09 looks like this:
 
; SIP Configuration for Asterisk;[general]port=5070   ; Port to bind tobindport=5070disallow=all    ; Disallow all codecsallow=ulawallow=alawallow=ilbc
allow=gsmdtmfmode=rfc2833relaxdtmf=yestos=lowdelay
context=myContextcanreinvite=nohost=dynamicinsecure=port,invitenat=yesqualify=1000
autocreatepeer=yes 
for the life of me, i cannot get SER to talk to asterisk1 (with 1.2 release). the same sip.conf doesn't work at all - asterisk completely ignores the requests (even at most verbose) - I've tried everything. if anyone has any ideas i'd be very grateful. I need to have SER talk to asterisk without defining a 
sip.conf entry for every entry.
 
really, i'm tearing my hair out - please help.
 
-yair
 
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RE: [asterisk-users] Asterisk -> Tekelec T6000 (Vocaldata, voiss)

[Watkins, Bradley]

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Matthew Crocker
> Sent: Thursday, September 28, 2006 3:20 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk -> Tekelec T6000 
> (Vocaldata, voiss)
> 
> 
> Thanks,
> 
>   The Tekelec T7000 is a traditional TDM class 4/5 switch 
> with VoIP interface cards (PIC) formerly known as the Taqua 
> OCX.  The Teklec T6000 is  a VoIP softswitch (feature server) 
> formerly known as the  
> VocalData VOISS.   I have both and I'm trying to get outbound calls  
> from a SIP phone registering with Asterisk through the T6000 
> to a T7000 and out to the PSTN.  Calls are working, DTMF is 
> not.  The T7000 is acting as the voice gateway to my T6000 
> and requires RFC2833.  So the Asterisk server has a sip.conf 
> that sends outbound calls to the T6000.  The T6000 is 
> configured to send 800# outbound to the T7000 which has 
> connectivity to the local Access Tandem and SS7 for IXC 
> termination.  The calls work fine, just can't navigate a 
> voice mail tree.
> 
> Tekelec doesn't officially support Asterisk, I have an open 
> ticket with them and I'm working on packet captures.  They 
> may be able to identify what is wrong with the config but 
> they won't be able to recommend fixes on the Asterisk side.
> 
> Anyone else have a T6000 working with Asterisk?
> 
> SIP signaling goes like this
> [SIP Phone] --> [Asterisk] --> [PIX FIrewall] --> [Tekelec 
> SBC] --> [T6000] --> [T7000 PIC]
> 
> Bearer traffic RTP goes like this
> 
> [SIP Phone] --> [PIX Firewall] --> [Tekelec SBC] --> [T7000 PIC]
> 
>  From my understanding RFC2833 means the DTMF is encoded in 
> the RTP stream so it is originating from the SIP phone,  
> Maybe the SIP phone is broken..  hrmm..
> 
> -Matt
> 
> 

Are you sure the RTP isn't going through the Asterisk box?  The reason I
ask is because this sounds suspiciously like the lack of variable-length
DTMF in pre-1.4 Asterisk (did you say what version of Asterisk you are
using and I missed it?).  Of course, depending on the phone, perhaps it
has a similar problem.

Regards,
- Brad
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Re: [asterisk-users] Sirrix to Legacy PBX

[Klaus Darilion]

Should work fine without problems. Sirrix has default NT wiring, thus 
you can use straight CAT5 cables.


regards
klaus

Marnus van Niekerk wrote:

Hi,

I have a site where we currently have * and a legacy PBX (Samsung DCS) 
both with BRI lines coming in.  The two in linked together with four 
analogue channels which works most of the time.


The local Telco in South Africa will (eventually) install a PRI with 
100DID's next week and we want to then use the existing sirrix board to 
feed 4BRI's to the legacy PBX effectively using it a a channel bank for 
the existing phones on that system.


Something like this:

PRI --> * --> sirrix 4BRI --> DCS

The DCS will think it is connected to the telco and make calls as usual 
(except for calls to internal extensions on * but that is a DCS 
programming issue.)



Anybody used a sirrix board like the before and any pointers before we 
attempt this?



Thank you


Marnus van Niekerk

--

"Opportunity is missed by most people because it is
dressed in overalls and looks like work."

Thomas Alva Edison - Inventor of 1093 patents,
including the light bulb, phonogram and motion pictures.




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[Asterisk-Users] Off-hooking Snom hanset doesn't answer incoming call

[Olivier]

Hi,Any advice on this one ?Sometimes off-hooking a Snom 320 handset doesn't answer a multi-extension call : phones keep ringing while handset remains silent.This occurs for 1 to 5% of incoming calls.
The setup is bristuffed 1.2.10 Asterisk with Junghanns QuadBRI and 6.2.x firmware Snom 320.Every call comes from ISDN network.Thanks in advance
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[asterisk-users] Re: max number of devices in hint

[Lacy Moore - Aspendora]

I have one extension that rings in many places.  It has just come to my attention that I can only monitor 4 devices within a hint.
 
Pretty sure I know what the problem is now.  It's not limited by devices, its limited by the length.  This is when it would be nice to know C.  I'm assuming the variables are declared and also declared as a certain length.

 
I need to increase this length for whatever variable holds the hint information.  Any hints? :-)  I don't even know where to start looking. 
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[asterisk-users] VMware and Digium TDM400P card

[Andy Green]

Title: VMware and Digium TDM400P card











Hello,

I have recently installed and upgraded trixbox in vmware on a win2000 server.

Everything works as expected but I can't seem to get vmware to see the card (it was installed after the vmware/trixbox was set up)

Should vmware be able to see the digium card automatically once zaptel is loaded, rebuilt, fresh installed etc.

Win2000 is asking for a PCI controller driver install, can this be ignored or do i really have to install some win drivers for the card, if so where do I find them?

Regards

Andy Green
IT Manager

GB eye Ltd
1 Russell St
Kelham Island
Sheffield
S3 8RW

Tel: 0114 252 1611
Fax: 0114 272 9599

mailto:[EMAIL PROTECTED]
http://www.gbeye.com





  







 



Coming Soon: WWE, Eregon (Fox Films), Torchwood (BBC), Iron Maiden, Terminator, Bill and Ted, Wedding Crashers, Muse, Reservoir Dogs (Prints & Canvas).



 


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Or browse our whole catalogue: http://www.gbeye.com/newsite/catalogue/

 


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Re: [asterisk-users] Re: max number of devices in hint

[Raphaël Jacquot]

Lacy Moore - Aspendora wrote:
> 
> I have one extension that rings in many places.  It has just come to
> my attention that I can only monitor 4 devices within a hint.
> 
>  
> Pretty sure I know what the problem is now.  It's not limited by
> devices, its limited by the length.  This is when it would be nice to
> know C.  I'm assuming the variables are declared and also declared as a
> certain length.
>  
> I need to increase this length for whatever variable holds the hint
> information.  Any hints? :-)  I don't even know where to start looking.

I noticed most strings in asterisk were of fixed length...
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[asterisk-users] Replacing mpg123 with madplay under Solaris?

[Frank Tarczynski]

I'm running Asterisk 1.2.12.1 on a Solaris 10 box.  I've built mpg123 
but it doesn't want to play well under Solaris so I want to replace it 
madplay.


I've edited app/app_mp3.c and res/res_musiconhold.c to change the calls 
for mpg123 to madplay with the appropriate options.


The madplay executable works find on this box from the command line but 
is giving a segmentation fault when called from Asterisk.


Has anyone already done this switch?  Can they share some pointers?

Frank

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[asterisk-users] Sip answer one side , ring other side

[antonio]

Hi, 

the scheme is this 
:
 
xlite  ---> 
Asterisk ---> SIP gateway  ---> PSTN
 
 
When i make a call 
with xlite (sip) to asterisk on the display of xlite i see that the call is 
connected but the phone is still ringing ..
What is the problem 
??
Thanks
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[asterisk-users] Polycom IP430 HUM and Echo

[Lorentz Hinrichsen]

I just implemented a Dell SC430, Sangoma A101, Adtran TA750 with 8FXO and 16FXS, and about 8 Poly IP430's.Everything is working pretty well, however users are complaining about the speakerphone quality and a loud HUM on the line.  I see in the 
sip.cfg that gains are set to all kinds of wacky numbers.  This is the official sip.cfg coming from the 2.01 firmware release.  It has quite a few specific lines for the IP430 - shouldn't these be flat?Thanks,
Wulf
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Re: [asterisk-users] Replacing mpg123 with madplay under Solaris?

[Matthias Fechner]

Hi Frank,

Frank Tarczynski schrieb:
> The madplay executable works find on this box from the command line but
> is giving a segmentation fault when called from Asterisk.
> 
> Has anyone already done this switch?  Can they share some pointers?

I run here Asterisk on FreeBSD with the buildin MOH from asterisk. It
plays here mp3s perfectly, why not use it too?


Best regards,
Matthias

-- 

"Programming today is a race between software engineers striving to
build bigger and better idiot-proof programs, and the universe trying to
produce bigger and better idiots. So far, the universe is winning." --
Rich Cook

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[asterisk-users] attended transfer unreliable

[Stefan Friedrich]

Hi,running asterisk 1.2.9 with freepbx 2.1.1, I have a strange problem:sometimes, call transfer works as expectet, and sometimes not. So far, I couldn't figure out any pattern in this behaviour,features.conf
:featuredigittimeout => 1500atxfer => *3-works:# user enters *Sep 29 14:52:14 DEBUG[21578] channel.c: Got DTMF on channel (SIP/230-a983)Sep 29 14:52:14 DEBUG[21578] 
channel.c: Bridge stops bridging channels SIP/210-859a and SIP/230-a983Sep 29 14:52:14 DEBUG[21578] res_features.c: Feature interpret: chan=SIP/210-859a, peer=SIP/230-a983, sense=2, features=18Sep 29 14:52:14 DEBUG[21578] res_features.c: Set time limit to 1500
Sep 29 14:52:14 VERBOSE[21578] logger.c: -- Attempting native bridge of SIP/210-859a and SIP/230-a983# user enters 3Sep 29 14:52:15 DEBUG[21578] channel.c: Got DTMF on channel (SIP/230-a983)Sep 29 14:52:15 DEBUG[21578] 
channel.c: Bridge stops bridging channels SIP/210-859a and SIP/230-a983Sep 29 14:52:15 DEBUG[21578] res_features.c: Feature interpret: chan=SIP/210-859a, peer=SIP/230-a983, sense=2, features=18# here is the transfer:
Sep 29 14:52:15 DEBUG[21578] res_features.c: Executing Attended Transfer SIP/210-859a, SIP/230-a983 (sense=2) XXXSep 29 14:52:15 VERBOSE[21578] logger.c: -- Started music on hold, class 'default', on SIP/210-859a
-doesn't work:# user enters *Sep 29 09:17:54 DEBUG[20534] channel.c: Got DTMF on channel (SIP/230-9e2a)Sep 29 09:17:54 DEBUG[20534] channel.c: Bridge stops bridging channels SIP/210-c701 and SIP/230-9e2a
Sep 29 09:17:54 DEBUG[20534] res_features.c: Feature interpret: chan=SIP/210-c701, peer=SIP/230-9e2a, sense=2, features=18Sep 29 09:17:54 DEBUG[20534] res_features.c: Set time limit to 1500Sep 29 09:17:54 VERBOSE[20534] 
logger.c: -- Attempting native bridge of SIP/210-c701 and SIP/230-9e2a#user enters 3Sep 29 09:17:55 DEBUG[20534] channel.c: Got DTMF on channel (SIP/230-9e2a)Sep 29 09:17:55 DEBUG[20534] channel.c: Bridge stops bridging channels SIP/210-c701 and SIP/230-9e2a
Sep 29 09:17:55 DEBUG[20534] res_features.c: Feature interpret: chan=SIP/210-c701, peer=SIP/230-9e2a, sense=2, features=18# no transferSep 29 09:17:55 DEBUG[17507] chan_sip.c: Stopping retransmission on '
[EMAIL PROTECTED]' of Request 102: Match FoundSep 29 09:17:55 DEBUG[17507] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]
' of Request 103: Match FoundSep 29 09:17:55 VERBOSE[20534] logger.c: -- Attempting native bridge of SIP/210-c701 and SIP/230-9e2a--when we have a timeout, it looks different:
Sep 29 12:00:34 DEBUG[21122] channel.c: Got DTMF on channel (SIP/240-6746)Sep 29 12:00:34 DEBUG[21122] channel.c: Bridge stops bridging channels Zap/2-1 and SIP/240-6746Sep 29 12:00:34 DEBUG[21122] res_features.c: Feature interpret: chan=Zap/2-1, peer=SIP/240-6746, sense=2, features=18
Sep 29 12:00:34 DEBUG[21122] res_features.c: Set time limit to 1500Sep 29 12:00:36 DEBUG[21122] channel.c: Got DTMF on channel (SIP/240-6746)Sep 29 12:00:36 DEBUG[21122] channel.c: Bridge stops bridging channels Zap/2-1 and SIP/240-6746
Sep 29 12:00:36 DEBUG[21122] res_features.c: Timed out for feature!hope your can help meStefan
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Re: [asterisk-users] VMware and Digium TDM400P card

[Jamin W. Collins]

Andy Green wrote:


I have recently installed and upgraded trixbox in vmware on a win2000 
server.


So, you have VMWare running on Windows 2000 Server (host) and are trying 
to run trixbox within a VMWare session (guest), do I follow you correctly?


Everything works as expected but I can't seem to get vmware to see the 
card (it was installed after the vmware/trixbox was set up)


I don't believe VMWare can provide the guest physical access to 
non-standard hardware (the TDM400P in this case).


In short the configuration you're attempting won't work, at least not in 
any way that I know of.


--
Jamin W. Collins
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Re: [Asterisk-Users] Off-hooking Snom hanset doesn't answer incoming call

[Klaus Darilion]

Olivier wrote:

Hi,

Any advice on this one ?

Sometimes off-hooking a Snom 320 handset doesn't answer a 
multi-extension call : phones keep ringing while handset remains silent.

This occurs for 1 to 5% of incoming calls.

The setup is bristuffed 1.2.10 Asterisk with Junghanns QuadBRI and 6.2.x 
firmware Snom 320.

Every call comes from ISDN network.


If the phone keeps ringing I guess it is a bug in the phone. I had a 
snom which had a broken switch (which detects off/on hook).


regards
klaus
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Re: [asterisk-users] extensions.conf strangeness

[Eric \"ManxPower\" Wieling]

Not having a [globals] section (even if it is empty) has caused Asterisk 
to screw things up in the past.  I think it causes contexts to not be found.


Brian Candler wrote:

On Thu, Sep 28, 2006 at 09:44:07AM -0500, Eric ManxPower Wieling wrote:

You need the [general] and [global] sections


Well, if you read the attachment, you would see that I had a [general]
section:

[general]
autofallthrough=no

In what way would adding an empty [global] section alter the anomoly I am
seeing?

Thanks,

Brian.



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Re: [asterisk-users] Sip answer one side , ring other side

[Brodie Macleod]

Try:

progressinband=no

in your sip.conf.

-Brodie

On Friday 29 September 2006 08:07 am, antonio wrote:
> Hi,
> the scheme is this :
>
> xlite  ---> Asterisk ---> SIP gateway  ---> PSTN
>
>
> When i make a call with xlite (sip) to asterisk on the display of xlite i
> see that the call is connected but the phone is still ringing ..
> What is the problem ??
> Thanks
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Re: [asterisk-users] t1-pri or sip trunk?

[Eric \"ManxPower\" Wieling]

stan ford wrote:

if you have to setup an office of 100 users now. would you rather setup a sip 
trunk,a t1-pri, or even a t1? and why?


Always a PRI.  PRIs have fast call setup, are reliable and work well.
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[asterisk-users] Extension Numbering

[Norbert Zawodsky]

Hi again,

as I wrote before, I'm new to Asterisk. And so, many many new questions
pop up .
For example:

I have here a very small telephony system. We have only 5 (or so)
extensions. (4 phones, 1 fax).
So I wonder if there is disadvantage if we use only 1 digit extensions
("1" for boss, "2" for shop, "4" for the fax ...)

I think I can remember something from the "Asterisk-TFOT" book saying
that one must not use 1-digit extension numbers. But I can't remember
that very well and can't find it in the book any more

Regards,
norbert

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Re: [asterisk-users] Re: Voip Buster - CID

[sip]

On Thu, 28 Sep 2006 21:33:13 -0400, Jay R. Ashworth wrote
> On Thu, Sep 28, 2006 at 11:31:29AM -0500, Eric ManxPower Wieling wrote:
> > Naija Man wrote:
> > >You can try VoipJet (http://www.voipjet.com)
> > >
> > >A simple configuration in you extensions.conf as below will solve your
> > >problem.
> > >
> > >exten => _X.,1,SetCIDNum(1341212)
> > >exten => _X.,n,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
> > 
> > "1" is not valid as the first digit in a NANPA phone number.  Only put 
> > the area code and phone number in the Caller*ID Information,
> 
> This seems like a good time to remind people of the difference 
> between a Directory Number, and what you dial to get to it.
> 
> The dialling pattern for NANP phone numbers usually includes the
> leading 1 (though some carriers let you leave it off), but the
> Directory Number doesn't include 1 -- as that is viewed as a
> non-international address.
> 
> Now, if you treat as a full E.164 address, it *does* include the 1,
> since that's the Country Code for the US.  But that's (usually) not
> germane to dialplanning.  I think.
> 

What, then, is the proper way of using full E164 numbers if not in the
dialplan and if not with Set(CALLERID(num)) ? 

N.

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[asterisk-users] [Fwd: asterisk-users Digest, Vol 26, Issue 166]

[asterisk-user]

I tried by adding "answer()" to the dial plan but the problem still exists.
I am not sure if I am doing this right.
Attached is the log file from asterisk while making the call to the conf 
bridge after adding "answer()"

Could you please let me know if you find anything out of this log file?

thanks for your help.

 Original Message 
Subject:asterisk-users Digest, Vol 26, Issue 166
Date:   Thu, 28 Sep 2006 07:42:43 -0700 (MST)
From:   [EMAIL PROTECTED]
Reply-To:   asterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com



Message: 19
Date: Thu, 28 Sep 2006 10:30:25 -0400
From: "BJ Weschke" <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] unable to call AT&T audio conference
bridge
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

On 9/28/06, asterisk-user <[EMAIL PROTECTED]> wrote:

Hello,
I have a problem with asterisk and trying to see if someone can help me
fix the issue...

Problem:
I couldn't join AT&T's Tele Conference bridge directly without their
customer service interaction.
Instead of getting the automated prompts to join the conference, it
takes me to the customer support and then I got to give them the bridge
number and pincode to add me into the conference call.

The reason given by AT&T was that their conference system is unable to
identify our tone.
This happens only with AT&T conference bridges... not sure what the
problem is.

This problem started after I installed trixbox on a new hardware.
Previous setup with [EMAIL PROTECTED]  did not have
this issue and I even switched back to [EMAIL PROTECTED]
 (a different box) and called the same conf
bridge... that worked fine.

I am running trixbox with the following versions:
asterisk - 1.2.9.1
zaptel - 1.2.8
libpri - 1.2.3-1.349
using zap over a 8 channel pri

Thanks in advance.



AT&T's IVR to collect the passcode is coming through as "early media"
and since you haven't signaled to the phones that the phone is
"answered" they're probably not letting you send DTMF through the
bridge that isn't technically supposed to be there yet.

Put an Answer() in your dial plan prior to sending the call out to
the Dial() application to reach the bridge for these types of calls
and this generally fixes your problems caused by someone else not
signaling correctly.

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/


--



Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Macro'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Expression result is '1'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'GotoIf'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Macro'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Expression result is '0'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'GotoIf'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Not taking any branch
Sep 28 19:30:04 DEBUG[32329] pbx.c: Expression result is '0'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'GotoIf'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Not taking any branch
Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is '208'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Set'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'NoOp'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is '208'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Set'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is ''
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Set'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Expression result is '0'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'GotoIf'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Not taking any branch
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Set'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is '"" <208>'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'NoOp'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is '208'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Macro'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is '0'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'GotoIf'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'AGI'
Sep 28 19:30:04 DEBUG[32330] app_queue.c: Device 'SIP/208' changed to state '2' 
(In use) but we don't care because they're not a member of any queue.
Sep 28 19:30:04 VERBOSE[32329] logger.c:   
recordingcheck|20060928-193004|1159486204.0: Outbound recording not enabled
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'NoOp'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Macro'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Expression result is '1'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'GotoIf'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'NoOp'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is ''
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Set'
Sep 28 19:30:04 DEBUG[32329] db.c: Unable to find key '208/emergency_cid' in 
family 'DEVICE'
Sep 28 19:30:04 DEBUG[32329] func_db.c: DB: DEVICE/208/emergency_cid not found 
in

RE: [Asterisk-Users] Off-hooking Snom hanset doesn't answer incom ing call

[Colin Anderson]

I got a bad batch of 360's where the hookswitch was damaged in shipping.
Snom fixed this by sticking a piece of packing foam between the switch and
the hook socket, wedging it into place. While this worked fine, I found I
had to be careful unpacking the phone - if you just yanked on the foam, a
small piece would tear off, wedging between the hookswitch and the socket.
Then the hookswitch would become intermittent. Take the phone apart - it's
not hard - and look for obstructions between the hookswitch and the frame. 

-Original Message-
From: Klaus Darilion [mailto:[EMAIL PROTECTED]
Sent: Friday, September 29, 2006 7:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Off-hooking Snom hanset doesn't answer
incoming call


Olivier wrote:
> Hi,
> 
> Any advice on this one ?
> 
> Sometimes off-hooking a Snom 320 handset doesn't answer a 
> multi-extension call : phones keep ringing while handset remains silent.
> This occurs for 1 to 5% of incoming calls.
> 
> The setup is bristuffed 1.2.10 Asterisk with Junghanns QuadBRI and 6.2.x 
> firmware Snom 320.
> Every call comes from ISDN network.

If the phone keeps ringing I guess it is a bug in the phone. I had a 
snom which had a broken switch (which detects off/on hook).

regards
klaus
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Re: [asterisk-users] Set hint status from dialplan?

[Andrew Kohlsmith]

On Tuesday 26 September 2006 12:12, C F wrote:
> IIRC, there was a dev status for the local channel being worked on the
> bug tracker.
> Ok, here is the link:
> http://bugs.digium.com/view.php?id=5779

Yes, but unless there is a way of setting a local channel's state, there's no 
way to achieve what Lacy Moore has done with a pristine checkout... is there?

-A.
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[asterisk-users] What minimum required packages for 1.4

[news.gmane.org]

I have just loaded Fedora Core 4 (minimum installation / updated).  Does 
anyone have an updated list of required packages or dependencies that need 
to be installed prior to basic (no real-time DB) Asterisk / Zaptel / libpri 
1.4 Beta install?  This is what I have so far and have used in the past 
w/1.2.x:


. glibc-devel
. ncurses-devel
. openssl-devel
. zlib-devel
. gcc
. gcc-c++
. kernel-devel
. bison

Does anyone have anything new to add to this?

Thanks,
LJ 



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Re: [asterisk-users] Sip answer one side , ring other side

[antonio]

It's the 
same
Thanks
 
 

Try:
progressinband=no
in your sip.conf.
-Brodie
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RE: [asterisk-users] t1-pri or sip trunk?

[Colin Anderson]

Also with PRI:

-Fax works
-No 911 issues
-SIP provider may or may not honor your arbitrarily set caller ID - PRI
always will if your telco isn't a dick
-Easier to "break out" an analog channel if needed (give me a channel bank
over an ata any day)
-Faster to troubleshoot - if you get red alarm there are only 2 things to
check - on SIP service, is it the LAN, firewall, switch, nic, IP gateway, or
unknown router on the internet? 

-Original Message-
From: Eric "ManxPower" Wieling [mailto:[EMAIL PROTECTED]
Sent: Friday, September 29, 2006 8:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] t1-pri or sip trunk?


stan ford wrote:
> if you have to setup an office of 100 users now. would you rather setup a
sip trunk,a t1-pri, or even a t1? and why?

Always a PRI.  PRIs have fast call setup, are reliable and work well.
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Re: [asterisk-users] Replacing mpg123 with madplay under Solaris?

[Ken Godee]

I'm running Asterisk 1.2.12.1 on a Solaris 10 box.  I've built mpg123 
but it doesn't want to play well under Solaris so I want to replace it 
madplay.


I've edited app/app_mp3.c and res/res_musiconhold.c to change the calls 
for mpg123 to madplay with the appropriate options.




I'm using a little older version of asterisk with Madplay, but
is it not still configured the same way, through the..

"musiconhold.conf"

;
; Music on hold class definitions
;
[classes]

default => custom:/var/lib/asterisk/mohmp3/,/usr/local/bin/madplay -Q 
--attenuate=-5 --mono -R 8000 --output=RAW:-


iron2 => custom:/var/lib/asterisk/iron2/,/usr/local/bin/madplay -Q -z 
--attenuate=-25 --fade-in --mono -R 8000 --output=RAW:-


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RE: [asterisk-users] pstn failback

[Shawn Kelley]

Stan,
I agree with the comment below, we switched from analog lines to a PRI and
it's not always as reliable as some people think. We are in a somewhat rural
location and we have outages regularly. 1-4 hour outages every few months
are not uncommon for us. Outages of 60 seconds or so are even more common.
I'm told this is because the T1 line is running somewhat noisy/dirty and
after so many CRC errors the equipment is resetting.

Make sure you negotiate a good SLA so that you can get credit when it does
go down!

You also have to be careful like mentioned below, if you get 2 PRI's, even
from different CLECS, the will normally still come out of the same Central
Office and travel side by side on the cable. So it's likely if 1 goes down
then the other one will also. 

Summary:  It's a good idea to have a few analog lines since they can take a
whole lot more abuse than the digital T1 can handle. (Static/Noise doesn't
make your call drop!)



-Original Message-
From: Lacy Moore [mailto:[EMAIL PROTECTED] 
Sent: Friday, September 29, 2006 4:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] pstn failback

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

stan ford wrote:
> On fonalities web page, i see they offer pstn failback as a feature of
their asterisk package. i've also heard before of failing back to a pri line
if your t1 voip line fails. my question is. in order to have pstn or pri
failback, dont you basically have to have all the equipment there on
standby, a PRI line, TDM cards, PRI/T1 cards, a bunch of digital or analong
phones. it just seems like a whole lot of hardware to be sitting there
waiting for a disaster. unless im just not understanding pstn/pri failback.
can someone shed some light?
>
>   also, if you've got a dedicated full t1 line for voice, and have a low
amount of users for that t1, is there really to worry about failing back to
a pstn? seeing how reliable a t1 is. is anyone out here using full voip
telelphony solution only?
>
Having had our XO connection go down within a week or so of switching to
a PRI, I can see how having a fallback would be good.  It was down for
about 4 hours.  However, that was back in May or June and hasn't been
down since.  I couldn't justify having something on standby for our
business.  But, our clients can reach us by phone in the office or cell,
and we can easily make outgoing calls on our cellphones. Our type of
business is not really dependent on absolutely having the phones up 100%
of the time.

If you do get a fallback, get it from different providers.  If your T1
from provider X is down, chances are a PRI from provider X will be
following the same path and be down as well.
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Version: GnuPG v1.4.5 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

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YwcV6yYYJX2cVly2z0dsdZ4=
=+Ik2
-END PGP SIGNATURE-



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[asterisk-users] automatic dialer with number pool and auto dialing

[Sam Tam]

Can I find out whether it is possible to achieve this.


"I would also like them to work with a dialler, cutting out Voicemails, no
answers and out of service calls giving them only calls that are live

With the dialer Ideally I would input a group of numbers into some type of
system (i.e excel) and it work through the numbers giving the advisors the
live calls!!"


Sam 


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[asterisk-users] real time billing system

[Pato Valarezo]

Hi, sorry for the question, i've been searching for a real time billing 
system for asterisk with zap/sip support, for use in post paid systems 
like "locutorios", do you know of or use any ?


thanks

--
patoVala
Linux User#280504
Hablando en http://www.elprimoalcahuete.com
"El tiempo cura los dolores y las querellas porque cambiamos. Ya no 
somos la misma persona. -- Blaise Pascal. (1600-1662) Filósofo y 
escritor francés. "

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RE: [asterisk-users] Good Book on Asterisk

[Race Vanderdecken]

The O'reilly Book "Asterisk, the Future of Telephony" is a good book.

I don't operate Asterisk; I just mess with the internal code and stuff
like that. This book helps me configure and understand how asterisk
works from the user side.

Race Vanderdecken
Code Tyrant
[EMAIL PROTECTED]
828 221 2636 vonage
828 699 2361 cel
Somewhere near Asheville, NC.
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Norbert
Zawodsky
Sent: Thursday, September 28, 2006 12:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Good Book on Asterisk

Michel Vaillancourt wrote:
> Norbert Zawodsky wrote:
>   
>> Hi everybody!
>>
>> I have some Linux experience but I'm completely new to asterisk.
>>
>> I bought a small VoIP-PBX which has Linux (Kernel 2.6.13) & Asterisk
>> (1.2.12) preinstalled and some basic configuration (Wiht a few
>> extensions). Now I want to implement something more, fox example
>> voicemail (storing voicemail data in an extern mysql DB) and so on.
>>
>> And since I don't want to waste your time with stupid questions 
>> ... can someone of you recommend a really good book on Asterisk? (To
buy
>> or for download)
>> ... or another online source of information which would be helpful
for
>> someone like me?
>>
>> I searched Amazon with "Asterisk" and got 21 hits..
>>
>> Thanks
>> Norbert
>>
>> 
>
> Hi, Norbert ... The O'Reily Book for Asterisk:
>
> http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
>
> Enjoy!
>   
Thanks for the book. I read it last night (or nearly all of it) and now
i think i understand a bit more.
But now the next question:

Where can I find the documentation of the applications and functions I
can use in the dialplan? (For example how to use the mysql add-on, ...)

Thanks
Norbert
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Re: [asterisk-users] Set hint status from dialplan?

[C F]

I have no clue that was just a refference, you tell me.

On 9/29/06, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote:

On Tuesday 26 September 2006 12:12, C F wrote:
> IIRC, there was a dev status for the local channel being worked on the
> bug tracker.
> Ok, here is the link:
> http://bugs.digium.com/view.php?id=5779

Yes, but unless there is a way of setting a local channel's state, there's no
way to achieve what Lacy Moore has done with a pristine checkout... is there?

-A.
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Re: [asterisk-users] real time billing system

[Chapeti]

Hola pato, hasta donde yo sé no hay nada que sea código abierto, lo que me 
parece mas fácil es que te hagas
uno propio, echale una ojeada a lo que hay en 
http://www.voip-info.org/wiki-Asterisk+manager+API, lo
único que haría falta sería un poco de conocimientos deVB 6 y de como 
trabajar con sockets ( cosa que no es nada del otro mundo ).


Saludos.



Hi, sorry for the question, i've been searching for a real time billing 
system for asterisk with zap/sip support, for use in post paid systems 
like "locutorios", do you know of or use any ?


thanks

--
patoVala
Linux User#280504
Hablando en http://www.elprimoalcahuete.com
"El tiempo cura los dolores y las querellas porque cambiamos. Ya no somos 
la misma persona. -- Blaise Pascal. (1600-1662) Filósofo y escritor 
francés. "

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[asterisk-users] Any Suggestions for Election Polling Application?

[Gerald Drouillard]

Looking to set up an outbound only Asterisk installation for 5 to 10 
attendants that will cold calling phone numbers in a database.  The 
customer would like the server to call the numbers as needed and 
transfer the call to an open attendant if a voice response is detected.


The customer called this "call banking" but it does not seem to 
translate directly into what Asterisk calls it?


Would Asterisk be able to do this?

Anybody have good experiences with softphone software?

Would Asterisk able to tranfer the person's name/phone number back to 
the softphone once the connection is made?


Any suggestions for SIP phones?

Any trouble with using ITSP like Vonage if the user has a good internet 
connection?


--
Regards
--
Gerald Drouillard
Technology Architect
Drouillard & Associates, Inc.
http://www.Drouillard.ca
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Re: [asterisk-users] real time billing system

[Bernardo Vieira]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Have you tried this:

http://www.voip-info.org/wiki-Asterisk+Prepaid+Applications

Pato Valarezo wrote:
> Hi, sorry for the question, i've been searching for a real time billing
> system for asterisk with zap/sip support, for use in post paid systems
> like "locutorios", do you know of or use any ?
> 
> thanks
> 

- --
"What most profoundly divides two men is a different sense and degree of
cleanliness. What help is all honesty and mutual utility, what help is
all the good will for each other: in the end the fact remains-they can't
stand each other?s smell!"

- - Nietzsche
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.1 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFFHU2G2QVs8jsa1mQRAjvkAJ4/xbuI8o54vjSDOSsFFzc/oASJCACghgGq
OgEKDq8jFQF/LWRSIHIPtr0=
=dSOU
-END PGP SIGNATURE-
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Re: [asterisk-users] rtc: lost some interrupts at 1024 when loading ztdummy

[Paul Hewlett]

On Tuesday 26 September 2006 05:26, Geoff Karl wrote:
> I am running today's SVN of the 1.4 branch, on Ubuntu dapper.
>
> I compiled a custom kernel (2.6.15.7).  Created modules of the rct and
> the rtc modlue loads fine.
>
> As soon as I load ztdummy the syslog fills up with:
>
> rtc: lost some interrupts at 1024 Hz.
>
> Any ideas what may be causing this?

I had a damaged motherboard that gave that error (the Pc had been dropped in 
transit).

Also, I recall that having the HPET timer enabled in th BIOS and/or the kernel 
may give this error. If the 'HPET timer supplies IRQ' is enabled in the 
kernel then rtc.ko is just a stub i.e. there is no actual code running. This 
will only be a problem if you are running just ztdummy.

PaulHewlett
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Re: [asterisk-users] Any Suggestions for Election Polling Application?

[Michel Vaillancourt]

Gerald Drouillard wrote:
> Looking to set up an outbound only Asterisk installation for 5 to 10
> attendants that will cold calling phone numbers in a database.  The
> customer would like the server to call the numbers as needed and
> transfer the call to an open attendant if a voice response is detected.
> 
> The customer called this "call banking" but it does not seem to
> translate directly into what Asterisk calls it?
> 
> Would Asterisk be able to do this?
> 
> Anybody have good experiences with softphone software?
> 
> Would Asterisk able to tranfer the person's name/phone number back to
> the softphone once the connection is made?
> 
> Any suggestions for SIP phones?
> 
> Any trouble with using ITSP like Vonage if the user has a good internet
> connection?
> 

This is exactly the application we are building right now.  Contact me 
offlist and I can put you in touch with one of our sales team.

-- 
--Michel Vaillancourt
Senior Telephony Engineer
Neoxo Inc  (www.neoxo.com)
+1 514 395 1106 ext 117
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[asterisk-users] SPA3000 register in asterisk

[Walter Willis]

I have like clients several spa 3000, problem is that spa3000 is not
registered or something by the east style problem must to be by
bandwidth? spa3000 verifies bandwidth qeu can use and that is
registered or no? very I am intrigued with this problemilla. Thanks your help.
	   
	   
	
 
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[asterisk-users] Asterisk IVR .wav issue

[David Moring]

  I have installed 
asterisk and TrixBox (newbie) and have configured it fine.  
However, when I try to upload a .wav file (8 khz mono) and use it for 
the primary IVR, nothing is played.  Also, when I try to locate 
the file on the box, I cannot find anything with the name I gave it 
(either .wav or .gsm).  Recording to an extension works fine 
  Any help or pointers much appeciated. 
   DMoring

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[asterisk-users] manager api redirect dropping calls

[Wing Wong]

Hi,I'm having some issues with the manager api when it tries to redirect a call.  If a call gets transferred to a person and the person doesn't answer, after the voicemail greeting the call gets dropped.  As well when I try to redirect a call to a queue, there is only one way audio.  If you need any more info let me know and I'll try to get you more details.
example:Action: RedirectChannel: SIP/10.3.14.253-b6b16c48Exten: 346Context: internalPriority: 1Thanks,Wing
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Re: [asterisk-users] VMware and Digium TDM400P card

[Joe Dennick]

VMWare specifically states that it doesn't provide guest access to any 
non-standard devices.  The guest operating systems only have access to 
the disks, network, sound card, and USB ports (if they are not already 
being used by another guest or host OS).  As such, VMWare is not a good 
solution for Asterisk if you need access to any Digium boards.  The 
trixbox configuration assumes that you are using only SIP and/or IAX 
connections to Asterisk.


Jamin W. Collins wrote:


Andy Green wrote:



I have recently installed and upgraded trixbox in vmware on a win2000 
server.



So, you have VMWare running on Windows 2000 Server (host) and are 
trying to run trixbox within a VMWare session (guest), do I follow you 
correctly?


Everything works as expected but I can't seem to get vmware to see 
the card (it was installed after the vmware/trixbox was set up)



I don't believe VMWare can provide the guest physical access to 
non-standard hardware (the TDM400P in this case).


In short the configuration you're attempting won't work, at least not 
in any way that I know of.



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RE: [asterisk-users] Any Suggestions for Election Polling Application?

[Dean Collins]

Hi Gerald,
What about initiating the calls inbound via the web? Check out
www.cognation.net/Mexuar 

We have also implemented a proof of concept using email to deliver the
same campaign mechanics as well.


Regards,

 

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]  
+1-212-203-4357 Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gerald
Drouillard
Sent: Friday, 29 September 2006 12:43 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Any Suggestions for Election Polling
Application?

Looking to set up an outbound only Asterisk installation for 5 to 10
attendants that will cold calling phone numbers in a database.  The
customer would like the server to call the numbers as needed and
transfer the call to an open attendant if a voice response is detected.

The customer called this "call banking" but it does not seem to
translate directly into what Asterisk calls it?

Would Asterisk be able to do this?

Anybody have good experiences with softphone software?

Would Asterisk able to tranfer the person's name/phone number back to
the softphone once the connection is made?

Any suggestions for SIP phones?

Any trouble with using ITSP like Vonage if the user has a good internet
connection?

--
Regards
--
Gerald Drouillard
Technology Architect
Drouillard & Associates, Inc.
http://www.Drouillard.ca
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Re: [asterisk-users] Any Suggestions for Election Polling Application?

[Matt Florell]

Several companies/organizations are doing this now with Asterisk and VICIDIAL.
http://astguiclient.sourceforge.net/vicidial.html

MATT---

On 9/29/06, Gerald Drouillard <[EMAIL PROTECTED]> wrote:

Looking to set up an outbound only Asterisk installation for 5 to 10
attendants that will cold calling phone numbers in a database.  The
customer would like the server to call the numbers as needed and
transfer the call to an open attendant if a voice response is detected.

The customer called this "call banking" but it does not seem to
translate directly into what Asterisk calls it?

Would Asterisk be able to do this?

Anybody have good experiences with softphone software?

Would Asterisk able to tranfer the person's name/phone number back to
the softphone once the connection is made?

Any suggestions for SIP phones?

Any trouble with using ITSP like Vonage if the user has a good internet
connection?

--
Regards
--
Gerald Drouillard
Technology Architect
Drouillard & Associates, Inc.
http://www.Drouillard.ca
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[asterisk-users] Re: Re: Set hint status from dialplan?

[Steven]

I guess the goal is to be able to give "any" (not just device or channel 
related) to a SIP subscription.

I can see many concepts that could use this:
Day/night mode.
Flash a light when nagios gets a red alarm.
Non phone related presence status. (IM, etc.)

I do not think that anyone would want to fake out real channel/device status.
The hint table exists, we just need a way to write and read it from the 
dialplan. (and prolly manager too)


-- 
-- 
Steven

http://www.glimasoutheast.org



"Hall, Eric M." <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
Here is an output from a 1.4.0-Beta2

voipgw*CLI> show channeltypes
TypeDescription  Devicestate
Indications  Transfer
--  ---  ---
---  
Agent   Call Agent Proxy Channel yes  yes
no
Console OSS Console Channel Driver   no   yes
no
Zap Zapata Telephony Driver w/PRIno   yes
no
Skinny  Skinny Client Control Protocol (Skinny)  no   yes
no
Phone   Standard Linux Telephony API Driver  no   yes
no
Feature Feature Proxy Channel Driver no   yes
no
SIP Session Initiation Protocol (SIP)yes  yes
yes
Local   Local Proxy Channel Driver   yes  yes
no
IAX2Inter Asterisk eXchange Driver (Ver 2)   yes  yes
yes
MGCPMedia Gateway Control Protocol (MGCP)yes  yes
no
--
10 channel drivers registered.
voipgw*CLI> show version
Asterisk 1.4.0-beta2 built by root @ voipgw on a i686 running Linux on
2006-09-25 00:49:44 UTC
voipgw*CLI>




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Tuesday, September 26, 2006 2:32 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Re: Set hint status from dialplan?

On Tuesday 26 September 2006 13:57, C F wrote:
> Andrew what does "show channeltypes" give you?

*CLI> show channeltypes
TypeDescription  Devicestate
Indications
Transfer
--  ---  ---
---

Zap Zapata Telephony Driver w/PRIno   yes

no
SIP Session Initiation Protocol (SIP)yes  yes

yes
Local   Local Proxy Channel Driver   yes  yes

no
IAX2Inter Asterisk eXchange Driver (Ver 2)   yes  yes

yes
Feature Feature Proxy Channel Driver no   yes

no
Agent   Call Agent Proxy Channel yes  yes

no
--
6 channel drivers registered.

*CLI> show version
Asterisk SVN-trunk-r41990 built by root @ asterisk on a i686 running
Linux on
2006-09-12 03:02:05 UTC

Curious... I see Local/ has a devicestate, and I've never heard of a
"Feature/" channel type before...  :-)

So I imagine I could use Local/[EMAIL PROTECTED] to see parking slot
state, but nothing for arbitrary channels such as what Lacy is showing.
Is that correct?

-A.
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Re: [asterisk-users] automatic dialer with number pool and auto dialing

[Matt Florell]

While no voicemail/answering machine-detection method is perfect, the
app_amd Asterisk module works pretty well if you tune it to your
setup.

The two Asterisk-based GPL Dialers both have this capability:
GnuDialer - http://www.gnudialer.org/
VICIDIAL- http://astguiclient.sourceforge.net/vicidial.html

MATT---


On 9/29/06, Sam Tam <[EMAIL PROTECTED]> wrote:

Can I find out whether it is possible to achieve this.


"I would also like them to work with a dialler, cutting out Voicemails, no
answers and out of service calls giving them only calls that are live

With the dialer Ideally I would input a group of numbers into some type of
system (i.e excel) and it work through the numbers giving the advisors the
live calls!!"


Sam


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Re: [asterisk-users] extensions.conf strangeness

[Michael Neuhauser]

On Thu, 2006-09-28 at 15:05 +0100, Brian Candler wrote:
> I have an anomoly that I am unable to explain.
> ...
>
> [invalid]
> exten => _X!,1,Answer()
> exten => _X!,2,Background(pbx-invalid)
> 
> [test]
> exten => 611,1,Answer()
> exten => 611,2,Playback(hello-world)
> exten => 611,3,Hangup()
> 
> [internal]
> include => extensions
> include => outbound
> include => invalid
> include => test
> 
> [from-sip]
> include => extensions
> include => outbound
> include => invalid
> include => test
> 
> ...
> [summary of cut text: 611 from SIP -> invalid; 611 from zaptel line -> test]
>
>I am running Asterisk from SVN trunk, compiled two weeks ago (September
>13th)

The order of include statements is important in 1.2, I don't know if
this still holds for trunk/1.4. Could you please try to include the
'invalid' context as the last one (i.e., AFTER "include => test", not
before) in both internal and from-sip and then test again?
-- 
Dr. Michael Neuhauser  mailto:[EMAIL PROTECTED]
Firmix Software GmbH  sip:[EMAIL PROTECTED]
Vienna/Austria/Europe   tel:+43-1-7890849-30
Linux Development and Services http://www.firmix.at/

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RE: [asterisk-users] pstn failback

[stan ford]

thanks for the responses     a couple things, if you guys could clear up for me.     A) If i have a bunch of sip ip phones, and if i were to fail to my PRI. I should still be able to use my ip phones right? i assume the signal will be changed by my T1 card? and the reverse i would assume is true for incoming calls.     B) How about DID's, how would that be handled. is there a DID failover as well? I have my VOIP service with one company, if i had my PRI service with another. how would those DID's get failed to the other provider, if thats even possible at all in a timely manner.     C) also are failover pri's generally cheaper that their active counterparts?     thanks alot.Shawn Kelley <[EMAIL PROTECTED]> wrote:  Stan,I agree with the comment below, we switched from analog lines to a PRI andit's not always as reliable as some people think. We are in a somewhat rurallocation and we have outages regularly. 1-4 hour outages every few monthsare not uncommon for us. Outages of 60 seconds or so are even more common.I'm told this is because the T1 line is running somewhat noisy/dirty andafter so many CRC errors the equipment is resetting.Make sure you negotiate a good SLA so that you can get credit when it doesgo down!You also have to be careful like mentioned below, if you get 2 PRI's, evenfrom different CLECS, the will normally still come out of the same CentralOffice and travel side by side on the cable. So it's likely if 1 goes downthen the other one will also. Summary: It's a good idea to have a few analog lines since they can take awhole lot more abuse than the digital T1 can handle. (Static/Noise
 doesn'tmake your call drop!)-Original Message-From: Lacy Moore [mailto:[EMAIL PROTECTED] Sent: Friday, September 29, 2006 4:23 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] pstn failback-BEGIN PGP SIGNED MESSAGE-Hash: SHA1stan ford wrote:> On fonalities web page, i see they offer pstn failback as a feature oftheir asterisk package. i've also heard before of failing back to a pri lineif your t1 voip line fails. my question is. in order to have pstn or prifailback, dont you basically have to have all the equipment there onstandby, a PRI line, TDM cards, PRI/T1 cards, a bunch of digital or analongphones. it just seems like a whole lot of hardware to be sitting therewaiting for a disaster. unless im just not understanding pstn/pri failback.can someone shed some light?> > also, if you've got a
 dedicated full t1 line for voice, and have a lowamount of users for that t1, is there really to worry about failing back toa pstn? seeing how reliable a t1 is. is anyone out here using full voiptelelphony solution only?> Having had our XO connection go down within a week or so of switching toa PRI, I can see how having a fallback would be good. It was down forabout 4 hours. However, that was back in May or June and hasn't beendown since. I couldn't justify having something on standby for ourbusiness. But, our clients can reach us by phone in the office or cell,and we can easily make outgoing calls on our cellphones. Our type ofbusiness is not really dependent on absolutely having the phones up 100%of the time.If you do get a fallback, get it from different providers. If your T1from provider X is down, chances are a PRI from provider X will befollowing the same path and be down as well.-BEGIN PGP
 SIGNATURE-Version: GnuPG v1.4.5 (MingW32)Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.orgiD8DBQFFHOX9VVXe/Qmwk9QRArSYAKDJN6Tf/L+L3ruXyXYcAeVbIyMxBwCgi2wMYwcV6yYYJX2cVly2z0dsdZ4==+Ik2-END PGP SIGNATURE-___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users 
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RE: [asterisk-users] Cisco CAll Manger and H323

[Dan Austin]

Yusuf wrote:
> Hi Dan,

> I used asterisk 1.2.10 with asterisk-addons 1.2.3.
> I did two successfull calls, but with dtmf=rfc2833, dtmf was not 
> sending at all.  Then when I made some changes, I could not get any
> calls to go through.  The call would just hangup after first ring.
Call Manager's support for RFC2833 is 'lacking'.  It works reasonably
well in 5.X for SIP, but forget about using it with H323.  I've tested
all four options with Call Manager, and only q931keypad and h245signal
worked.  I'd recommend using h245signal.

> Did you get calls going in both ways, inbound and outbound to
asterisk.
>  O got two calls going from CAllmanger to asterisk only, other would
not
> work.

Calls work both ways, although 99.99% of my calls are inbound, since
we use Asterisk for conferencing only at this point.

Here are a couple of ideas to try:
1.  Set the Call Manager H323 gateway to 'Require MTP'
2.  Set DTMF to h245 signal

What is likely happening is that with Asterisk asking for RFC2833,
CCM tries to invoke a MTP.  I am not sure in which Asterisk-Addons
version it was added, but I wrote support for Empty Terminal Capability
sets for chan_ooh323.  If that feature is not in the version you have,
(chan_ooh323 release 0.5 or newer), and you are not forcing an MTP on
the CCM gateway you'll see a problem like you have.

I should also point out that if you are not running chan_ooh323 0.5 or
newer and do get Asterisk to accept calls with out forcing an MTP, calls
will be dropped anytime a CCM endpoint uses hold or transfer features.

If you do have 0.5 or newer, then changing the DTMF migth be enough.

Hope this helps,
Dan
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Re: [asterisk-users] Asterisk IVR .wav issue

[David Moring]

  Oops, I did find them, but they 
are (correctly?) set to asterisk owner in custom...  Any ideas?
 David MoringMC 2188http://www.fleet4.com 
-Original 
Message-From: "David Moring" <[EMAIL PROTECTED]>To: 
asterisk-users@lists.digium.comDate: Fri, 29 Sep 2006 13:21:01 -0400
Subject: [asterisk-users] Asterisk IVR .wav issue I have installed 
asterisk and TrixBox (newbie) and have configured it fine.  
However, when I try to upload a .wav file (8 khz mono) and use it for 
the primary IVR, nothing is played.  Also, when I try to locate 
the file on the box, I cannot find anything with the name I gave it 
(either .wav or .gsm).  Recording to an extension works fine 
  Any help or pointers much appeciated. 
   DMoring

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[asterisk-users] Queue and Pickup/DPickup

[Ole Myhre]

Hi,

 

Is it possible to either pick up calls to phones that go
through a queue (using Pickup/DPickup), or not notify other users when phones
are ringing because of a queue?

 

When the call goes through a queue, there is no extensions
that is being dialed, and therefore just using DPickup(exten) wont work, at
least not here. Other users gets notified when phones are ringing, and its kind
of pointless to see a blinking light on peoples phones when users can’t
know if its possible to answer the call or not.

 

By the way, it works perfectly with calls going directly
through extensions.conf. J

 

Ole Myhre






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[asterisk-users] recommended application for salesman using asterisk

[Yu Safin]

Hi, I am a salesman currently using asterisk to contact my customers.
So far, I have asterisk connected to two PSTN analog lines where I
only receive phones calls.
Then, I have asterisk connected to a VoIP service company for
terminating my phone calls.
I also kept one PTSN phone line to place calls to my cellular when I
am on the road.  This is done by giving the caller an option to find
me on my cellular.
I started to tinker with my PC and I can now receive and place calls
using a soft-phone (iaxclient).  Then I started to wonder if there is
an application that will allow me to use a friendly WEB interface
along side with my soft-phone to quickly place phone calls from an
address book and when I call arrives, to bring up information about
the caller if present in my address book.
I also conceived the idea that I might not even have the imagination
that some members of this list may have in terms of how else I can be
exploiting asterisk e.g. callback from messages left.  Another one is
a more sophisticated find-me service that I can manage from a WEB
interface.
I am open to ideas and suggestions.  Please show me where to go.
Remeber, if I spend too much time as an engineer I go hungry.  I need
to be in front of people selling my company services and not tinkering
with applications.
Thank you.
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Re: [asterisk-users] pstn failback

[Lacy Moore - Aspendora]

 
a couple things, if you guys could clear up for me.
 
A) If i have a bunch of sip ip phones, and if i were to fail to my PRI. I should still be able to use my ip phones right? i assume the signal will be changed by my T1 card? and the reverse i would assume is true for incoming calls.

 
Outgoing could be handled by the dial plan.  Incoming would have to be something worked out with the providers.

 
B) How about DID's, how would that be handled. is there a DID failover as well? I have my VOIP service with one company, if i had my PRI service with another. how would those DID's get failed to the other provider, if thats even possible at all in a timely manner.

 
Not sure how this would be handled.  

 
C) also are failover pri's generally cheaper that their active counterparts?
 
thanks alot. 

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Re: [asterisk-users] pstn failback

[Jay R. Ashworth]

On Fri, Sep 29, 2006 at 10:39:44AM -0500, Shawn Kelley wrote:
> You also have to be careful like mentioned below, if you get 2 PRI's, even
> from different CLECS, the will normally still come out of the same Central
> Office and travel side by side on the cable. So it's likely if 1 goes down
> then the other one will also. 

Indeed.  Acquiring, and *maintaining*, physical diversity of backup
circuits is one of the thorniest problems in telecom management. 

IBM has a hard time getting carriers to get it right, so why should we
be any different?

Cheers,
-- jra
-- 
Jay R. Ashworth[EMAIL PROTECTED]
Designer  Baylink RFC 2100
Ashworth & AssociatesThe Things I Think'87 e24
St Petersburg FL USA  http://baylink.pitas.com +1 727 647 1274

"That's women for you; you divorce them, and 10 years later,
  they stop having sex with you."  -- Jennifer Crusie; _Fast_Women_
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Re: [asterisk-users] Building the Perfect Box

[Jay R. Ashworth]

On Fri, Sep 29, 2006 at 09:37:06AM +0100, Conrad Wood wrote:
> > > 7. Relationship with provider. What is their SLA? Is it the
> > > incumbent or the clec? An incumbent will be more expensive
> > > and more difficult to deal with but they will tend to be more
> > > reliable. A clec will be cheaper and they will be way more
> > > accomodating but you will most likely not get five 9's from them.
> > > A VoIP provider should never be trusted, period. You will not get
> > > five nines from them, ever. Plan failover situations accordingly.
> >
> > "SLA? What's a 'SLA'?" :-)
>
> Service Level Agreement. Normally it means if your line fails you
> get £50 or so. Maybe £100. But *never* enough to compensate for the
> trouble. You have to have a backup plan.

I'm sorry.  You seem to have fallen into the sar-chasm.

And I thought the smiley would be enough hint.  :-)

> > Amusingly, a client's * box went down this morning. I didn't get the
> > washout, but the mitigation wasn't well planned either -- everyone
> > with an Asterisk box should know what they're going to do if it
> > falls over, in detail. In a notebook. Just like when the nuclear
> > missles start going.
>
> Yes of course. These notebooks tend to get forgotten in a cupboard
> until the day they're needed. And then they're so out of date that
> they're more damaging that useful. Do you update the text on the
> server itself somewhere? How do you go about keeping it up-to-date?
> "Just discipline"? Do you work in a team with others?

I think it's possible to discipline yourself on this front, yes.  How
often does an operational *-box change enough that you have to modify
your recovery procedures, anyway?

Cheers,
-- jra
-- 
Jay R. Ashworth[EMAIL PROTECTED]
Designer  Baylink RFC 2100
Ashworth & AssociatesThe Things I Think'87 e24
St Petersburg FL USA  http://baylink.pitas.com +1 727 647 1274

"That's women for you; you divorce them, and 10 years later,
  they stop having sex with you."  -- Jennifer Crusie; _Fast_Women_
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[asterisk-users] Re: asterisk-users Digest, Vol 26, Issue 172

[Frank Tarczynski]

>
> Message: 9
> Date: Fri, 29 Sep 2006 08:23:29 -0700
> From: Ken Godee <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] Replacing mpg123 with madplay under
>   Solaris?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>   
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset=us-ascii; format=flowed
>
>> I'm running Asterisk 1.2.12.1 on a Solaris 10 box.  I've built mpg123
>> but it doesn't want to play well under Solaris so I want to replace it
>> madplay.
>>
>> I've edited app/app_mp3.c and res/res_musiconhold.c to change the calls
>> for mpg123 to madplay with the appropriate options.
>>
>
> I'm using a little older version of asterisk with Madplay, but
> is it not still configured the same way, through the..
>
> "musiconhold.conf"
>
> ;
> ; Music on hold class definitions
> ;
> [classes]
>
> default => custom:/var/lib/asterisk/mohmp3/,/usr/local/bin/madplay -Q
> --attenuate=-5 --mono -R 8000 --output=RAW:-
>
> iron2 => custom:/var/lib/asterisk/iron2/,/usr/local/bin/madplay -Q -z
> --attenuate=-25 --fade-in --mono -R 8000 --output=RAW:-
>

Yes, I have madplay configured in my musiconhold.conf file too.  But if I
leave mpg123 in the app_mp3.c and res_musiconhold.c files then asterisk
will spawn mpg123 processes as well.  On my Solaris box these mpg123
processes will slowly consume more and more of the CPU until killed. 
madplay is much better behaved.

I've basically just edited the app_mp3.c and res_musiconhold.c files
replacing the mpg123 entries with madplay.  My problem seems to be with
getting the options for madplay right so it will work when spawned from
inside asterisk.

Any help would be appreciated.

Frank


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Re: [asterisk-users] Forcing Transcode

[Mr. Jones]

Thanks -

This worked. I swear I was getting a 503 or something weird before
when I did this but it seems to be working now.

On 9/28/06, Andres <[EMAIL PROTECTED]> wrote:

Mr. Jones wrote:

> Hi Folks,
>
> I'm curious if there's anyway to force Asterisk to transcode for
> certain handsets.

All you need is to edit your sip.conf and force the codec per user entry:

[your entry here]
disallow=all
allow=g729

>
> Specifically we have an inbound SIP origination service which uses g711.
>
> We're having bandwidth issues with a client and would like to force
> Asterisk to transcode to g729 until we can get their T1 in place.
>
> Any ideas?
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>


--
Andres
Technical Support
http://www.telesip.net

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[asterisk-users] Problems with DISA

[Shidan]

For some reason I'm having problems with DISA. This is what I have:

exten => s,1,Answer()
exten => s,2,DigitTimeout(5)
exten => s,3,ResponseTimeout(10)
exten => s,4,DISA(no-password|from-internal)

I can generate tones with no problems on this system
But I cant hear a dialtone using DISA, anyone experienced
this problem? Also, I'm running this in the cusom confs for
freepbx.

Just to save time, I'm not looking for a solution that doesn't
use the Disa app.

Cheers,
Shidan Gouran
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[asterisk-users] A Step back with AddQueueMember() ?

[Douglas Garstang]

All,

The queue function AgentCallBackLogin() would take the name of an agent as an 
argument, and would log that agent into the queue they where associated with. 
We programmed our appearances on our Polycom phones to have a different 
appearance for each queue, and we'd send the caller id of the appearance as the 
name of the agent, thereby removing the need for the agent to enter their agent 
number. 

I see though that the AddQueueMember() function requires a queue name as an 
argument. That makes the dialplan logic more complex as we somehow have to 
magically send the queue name that this agent belongs to from the phone to the 
dialplan.

Doug.

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Re: [asterisk-users] Replacing mpg123 with madplay under Solaris?

[Tzafrir Cohen]

On Fri, Sep 29, 2006 at 08:47:27AM -0400, Frank Tarczynski wrote:
> I'm running Asterisk 1.2.12.1 on a Solaris 10 box.  I've built mpg123 
> but it doesn't want to play well under Solaris so I want to replace it 
> madplay.

Just stating the obvious: the only please where you'd need mpg123 is for
streaming mp3 from a remote site. And even then, there is a performance
penalty. If you have control of the streaming source, streaming raw
sinear, or gsm, can save much on the performance side.

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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Re: [asterisk-users] SIP Gateway

[Forrest Beck]

Not that server model, right now we have a dell server for testing
(which puts the server cost ata round $2000).  I am hoping to get one
in to see if it will play nice with the TDM cards.  This appealed to
me because of it's small rackable form factor and cheap price.  For
that price I can have a cold/hot spare.  I will post again if I have
luck.

On 9/26/06, Kevin Kiely <[EMAIL PROTECTED]> wrote:

Forrest,

I noticed your post on the mailing list and was curious if you had used that
server before with asterisk with any TDM cards in it?

Kevin



-Original Message-
From: Forrest Beck [mailto:[EMAIL PROTECTED]
Sent: Tuesday, September 26, 2006 1:50 PM
To: Asterisk Users List
Subject: [asterisk-users] SIP Gateway

I am thinking of using a mini atx 1u server with a digium zaptel
(wcte11xp) installed to act as a SIP gateway.  This way any of my
asterisk servers can forward calls to any gateway (seperated by about
3miles of fiber).   Has anyone else tried this?  I would just load a
basic asteisk config and zaptel with something like CentOS 4.4
ServerCD.  Here is the hardware I am thinking of.

http://www.abmx.com/1u-short-depth-rack-mount-server-p-256.html

It seems like this would be alot cheaper than getting a pre-built sip
gateway from VOX.

Any input is greatly appreciated.

Forrest
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Re: [asterisk-users] Problems with DISA

[Eric \"ManxPower\" Wieling]

Make sure you have a /etc/asterisk/indications.conf

Shidan wrote:

For some reason I'm having problems with DISA. This is what I have:

exten => s,1,Answer()
exten => s,2,DigitTimeout(5)
exten => s,3,ResponseTimeout(10)
exten => s,4,DISA(no-password|from-internal)

I can generate tones with no problems on this system
But I cant hear a dialtone using DISA, anyone experienced
this problem? Also, I'm running this in the cusom confs for
freepbx.

Just to save time, I'm not looking for a solution that doesn't
use the Disa app.

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RE: [asterisk-users] Cisco CAll Manger and H323

[Yusuf]

Thanks Dan,

that was awesome, and really made sense about what was really happening.  :)
Will try a newer.

BTW: I did get it to successfully route inbound calls to asterisk with
oh323, and DTMF and transfers worked fine.


> Yusuf wrote:
>> Hi Dan,
>
>> I used asterisk 1.2.10 with asterisk-addons 1.2.3.
>> I did two successfull calls, but with dtmf=rfc2833, dtmf was not
>> sending at all.  Then when I made some changes, I could not get any
>> calls to go through.  The call would just hangup after first ring.
> Call Manager's support for RFC2833 is 'lacking'.  It works reasonably
> well in 5.X for SIP, but forget about using it with H323.  I've tested
> all four options with Call Manager, and only q931keypad and h245signal
> worked.  I'd recommend using h245signal.
>
>> Did you get calls going in both ways, inbound and outbound to
> asterisk.
>>  O got two calls going from CAllmanger to asterisk only, other would
> not
>> work.
>
> Calls work both ways, although 99.99% of my calls are inbound, since
> we use Asterisk for conferencing only at this point.
>
> Here are a couple of ideas to try:
>   1.  Set the Call Manager H323 gateway to 'Require MTP'
>   2.  Set DTMF to h245 signal
>
> What is likely happening is that with Asterisk asking for RFC2833,
> CCM tries to invoke a MTP.  I am not sure in which Asterisk-Addons
> version it was added, but I wrote support for Empty Terminal Capability
> sets for chan_ooh323.  If that feature is not in the version you have,
> (chan_ooh323 release 0.5 or newer), and you are not forcing an MTP on
> the CCM gateway you'll see a problem like you have.
>
> I should also point out that if you are not running chan_ooh323 0.5 or
> newer and do get Asterisk to accept calls with out forcing an MTP, calls
> will be dropped anytime a CCM endpoint uses hold or transfer features.
>
> If you do have 0.5 or newer, then changing the DTMF migth be enough.
>
> Hope this helps,
> Dan
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thanks,
yusuf


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[asterisk-users] native sounds

[Ed Nuñez]

>From where can I download the collection of Asterisk Native Sounds?

I tried the www.astlinux.com link, but I was not able to uncompress them 
because they seem to be corrupted.  

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Re: [asterisk-users] Ericsson MD110

[Andrew Joakimsen]

Well if you currently have a T1/E1 just get a 2 port card and place it
inline to the PBX. All calls will go to asterisk so they can all be
billed and you can handle voicemail.

However the most complex issue would be voicemail notifications

On 9/26/06, Patricio A. Bruna <[EMAIL PROTECTED]> wrote:

Has anyone a workin setup between asterisk and an ericsson md110 pbx?
i need asterisk to do the billing and voicemail work, so i think it should
be connected directly to the pstn and pass the calls to the md110 and
viceversa.

any recomendations?

thx


Patricio Bruna V.
Red Hat Certified Engineer
IT Linux Ltda.
http://www.it-linux.cl
Fono : (+56-2) 333 0051
Cel  : (+56-09) 8288 5195

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Re: [asterisk-users] pstn failback

[Steve Glaus]

 
B) How about DID's, how would that be handled. is there a DID failover 
as well? I have my VOIP service with one company, if i had my PRI 
service with another. how would those DID's get failed to the other 
provider, if thats even possible at all in a timely manner.
I have yet to come up with any way that  failure of our DID provider 
could be handled. We've had to do a fair bit of shopping around before 
we found a voip provider that has been reliable (Voipstreet). They've 
had very good service to this point (about 3 months now. I don't know 
what we would do if they went out. A PRI is just far to expensive to 
consider for only backup purposes.


DID's have been somewhat of a problem for us. There aren't that many 
providers and those that exist are more expensive than I thought it 
would be (or if they're cheap their service is so unreliable as to be 
worthless). Maybe someone here can give me an idea of how they do their 
DID's ? We've tried didx.org as well but most of the numbers we've 
gotten from them have horrible audio problems. 

Maybe someone knows of a super reliable provider with unlimited 
incoming/ > 2 channels for about 10$ a month?


Termination is no problem. Lots of providers and the dialplan will 
automatically failover if one provider doesn't work.

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Re: [asterisk-users] ATA with wireless client

[Andrew Joakimsen]

The cheapest suggestion is to buy a Buffalo WHR-G54S, it is the same
as the GPL linksys routers, load the DD-WRT firmware and then you can
use it as a 5 port wireless Ethernet bridge, they cost less than USD
50.

On 9/22/06, Brian Candler <[EMAIL PROTECTED]> wrote:

Sorry, one other equipment query: does anyone know of an ATA with wireless
hardware which can act as a *client* to another wireless network?

The Linksys units have an integrated wireless access point, but I want
something which will work as a client onto an existing wireless network - so
you can install ATAs around a building without additional LAN cabling.

An ATA with integrated Homeplug (powerline carrier networking) would be
another option, but again I can't find such a thing.

Any suggestions?

Many thanks,

Brian.
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Re: [asterisk-users] DSL router with integrated SIP proxy?

[Andrew Joakimsen]

The VoIP version of DD_WRT runs Ser by default

On 9/24/06, David Gagnon <[EMAIL PROTECTED]> wrote:

You could take a WRTSL54gs, install openwrt then openser

David

-Message d'origine-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Steve Kennedy
Envoyé: 24 septembre 2006 08:47
À: asterisk-users@lists.digium.com
Objet: Re: [asterisk-users] DSL router with integrated SIP proxy?

On Thu, Sep 21, 2006 at 10:11:43PM +0100, Brian Candler wrote:

> Does anyone here know of an ADSL router with integrated SIP proxy?

Netscreen 5GT ADSL, it has what's called an ALG (application layer
gateway) and it does indeed support SIP. Full featured firewall etc too.


Steve

p.s Hi Brian :)

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[asterisk-users] Re: SIP Gateway

[Forrest Beck]

To rich for my blood. Googled it.  Looks like it is about $12000, I
hope to stay in the $1500 range.  We are but mearly a private school.

On 9/29/06, James <[EMAIL PROTECTED]> wrote:

I use the Lucent MAX TNT.

They are cheap, will do up to 24 T1's, have 12 fans and I've never had one
fail.
I also can't remember the last time that I had to reboot on of them.
G.711 & G.729 is built in.

James Taylor
1-903-691-0069

- Original Message -
From: "Forrest Beck" <[EMAIL PROTECTED]>
To: "Asterisk Users List" 
Sent: Tuesday, September 26, 2006 12:50 PM
Subject: [asterisk-users] SIP Gateway


>I am thinking of using a mini atx 1u server with a digium zaptel
> (wcte11xp) installed to act as a SIP gateway.  This way any of my
> asterisk servers can forward calls to any gateway (seperated by about
> 3miles of fiber).   Has anyone else tried this?  I would just load a
> basic asteisk config and zaptel with something like CentOS 4.4
> ServerCD.  Here is the hardware I am thinking of.
>
> http://www.abmx.com/1u-short-depth-rack-mount-server-p-256.html
>
> It seems like this would be alot cheaper than getting a pre-built sip
> gateway from VOX.
>
> Any input is greatly appreciated.
>
> Forrest
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Re: [asterisk-users] Problems getting 7970G upgraded to SIP

[Jason Lixfeld]

On 12-Sep-06, at 3:14 PM, Richard Klingler wrote:


Hi Jason


Hi!  Sorry for the delay! :/

SIP70.8-0-4SR1SloadInformation6>


1. Stick with the 8.0.2 SIP image as it works best with asterisk...
   at least for me (o;

- Here are TFTP server logs to illustrate that I'm using the  
correct case'd XmlDefault.cnf.xml file:
Sep 10 21:57:55 bubbles  tftpd[89195]: jalc7970.sip : read request  
for SEP00131A4D39F4.cnf.xml: File not found


2. I thought you created your SEP file? And still it can't be  
found?


the SEPxxx file IS found, but only if the XmlDefault.cnf.xml doesn't  
exist.  If the XmlDefault does exist, it never asks for the SEPxxx file.


Sep 10 21:57:55 bubbles  tftpd[89197]: jalc7970.sip : read request  
for //XmlDefault.cnf.xml: success


3. Wondering what messages are coming after that...or is it the
   point where it starts over again?


Correct, that's where it loops.

- All the files from the .cop are 100% unmodified.  I just tar - 
zxvf cmterm-7970_7971-sip.8-0-4SR1.cop and the files are extracted  
into the tftpd root directory, which is the same place the SEP and  
XmlDefault file are located.


4. So you have all those:

-bash-2.05b$ tar tzvf cmterm-7970_7971-sip.8-0-2SR1.cop
 644 Mar 22 23:49 SIP70.8-0-2SR1S.loads
 2538161 Mar 22 23:49 apps70.1-1-1-15.sbn
  411264 Mar 22 23:49 cnu70.3-1-1-15.sbn
1996 Mar 23 00:06 copstart.sh
 2401588 Mar 22 23:49 cvm70sip.8-0-1-18.sbn
  483105 Mar 22 23:49 dsp70.1-1-1-15.sbn
  465288 Mar 22 23:49 jar70sip.8-0-1-18.sbn
  71 Mar 23 00:06 load119.txt
  72 Mar 23 00:06 load30006.txt
   0 Mar 23 00:06 signed/
 4046848 Mar 23 00:06 signed/cmterm-7970_7971-sip.8-0-2SR1.cop
 644 Mar 22 23:49 term70.default.loads
 644 Mar 22 23:49 term71.default.loads


I'll give 8-0-2SR1 a shot:

[EMAIL PROTECTED] /tftpboot]# tar -zxvf cmterm-7970_7971-sip.8-0-2SR1.cop
x SIP70.8-0-2SR1S.loads
x apps70.1-1-1-15.sbn
x cnu70.3-1-1-15.sbn
x copstart.sh
x cvm70sip.8-0-1-18.sbn
x dsp70.1-1-1-15.sbn
x jar70sip.8-0-1-18.sbn
x load119.txt
x load30006.txt
x signed/
x signed/cmterm-7970_7971-sip.8-0-2SR1.cop
x term70.default.loads
x term71.default.loads
[EMAIL PROTECTED] /tftpboot]# grep 30006 XmlDefault.cnf.xml
SIP70.8-0-2SR1loadInformation30006>

[EMAIL PROTECTED] /tftpboot]# tail -f /var/log/tftpd.log
Sep 29 16:03:50 bubbles.  tftpd[67175]: jalc7970.sip. : read request  
for SEP00131A4D39F4.cnf.xml: File not found
Sep 29 16:03:51 bubbles.  tftpd[67177]: jalc7970.sip. : read request  
for //XmlDefault.cnf.xml: success
Sep 29 16:04:35 bubbles.  tftpd[67213]: jalc7970.sip. : read request  
for SEP00131A4D39F4.cnf.xml: File not found
Sep 29 16:04:35 bubbles.  tftpd[67215]: jalc7970.sip. : read request  
for //XmlDefault.cnf.xml: success

...
...
Sep 29 16:05:20 bubbles.  tftpd[67242]: jalc7970.sip. : read request  
for //SEP00131A4D39F4.cnf.xml: success


So, same thing.  if SEPxxx doesn't exist, XmlDefault.cnf.xml is found  
and loaded but no upgrade happens.


If SEPxxx does exist, I get LoadID Incorrect on the phone with no  
upgrade apparently taking place.




Anyone have any ideas?


5. Not yet. But might be you need to go with a firmware
   in between first before going with 8.0.x.


Interesting thought.  So I'm running TERM70.5-0-3-0S (assuming that's  
SCCP).  Maybe I have to go to 5.X SIP, then 8 or through 5-6-7-8 in  
there somehow?



cheers
rick


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[asterisk-users] Fiber Outage: LA->Washington

[Jay R. Ashworth]

Backbone folks are noting that a fiber cut went down on the west coast,
sometime after 0945PDT today, between Washington (I think that's
"State") and LA, on the order of 5 OC-48's.

So if you're having troubles with VoN today, that's probably why.

Cheers,
-- jra
-- 
Jay R. Ashworth[EMAIL PROTECTED]
Designer  Baylink RFC 2100
Ashworth & AssociatesThe Things I Think'87 e24
St Petersburg FL USA  http://baylink.pitas.com +1 727 647 1274

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  they stop having sex with you."  -- Jennifer Crusie; _Fast_Women_
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Re: [asterisk-users] Re: SIP Gateway

[Eric \"ManxPower\" Wieling]

Try eBay

Forrest Beck wrote:

To rich for my blood. Googled it.  Looks like it is about $12000, I
hope to stay in the $1500 range.  We are but mearly a private school.

On 9/29/06, James <[EMAIL PROTECTED]> wrote:

I use the Lucent MAX TNT.

They are cheap, will do up to 24 T1's, have 12 fans and I've never had 
one

fail.
I also can't remember the last time that I had to reboot on of them.
G.711 & G.729 is built in.

James Taylor
1-903-691-0069

- Original Message -
From: "Forrest Beck" <[EMAIL PROTECTED]>
To: "Asterisk Users List" 
Sent: Tuesday, September 26, 2006 12:50 PM
Subject: [asterisk-users] SIP Gateway


>I am thinking of using a mini atx 1u server with a digium zaptel
> (wcte11xp) installed to act as a SIP gateway.  This way any of my
> asterisk servers can forward calls to any gateway (seperated by about
> 3miles of fiber).   Has anyone else tried this?  I would just load a
> basic asteisk config and zaptel with something like CentOS 4.4
> ServerCD.  Here is the hardware I am thinking of.
>
> http://www.abmx.com/1u-short-depth-rack-mount-server-p-256.html
>
> It seems like this would be alot cheaper than getting a pre-built sip
> gateway from VOX.
>
> Any input is greatly appreciated.
>
> Forrest
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Re: [asterisk-users] Queue AddQueueMember()

[BJ Weschke]

On 9/28/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:

All,

I've recently been told that the AgentCallBacklogin() application is buggy, and 
I should not use it. Apparently I should use AddQueueMember() instead. I see 
though that AddQueueMember() does not take the location to call back as an 
argument.

We have remote agents that are available via PSTN access only. With 
AgentCallBackLogin() they can enter their PSTN phone number, and Asterisk will 
call them back at that number when they get a queue call. Can AddQueueMember() 
do that?

Is AgentCallBackLogin() going to be deprecated at some point? Will 
AddQueueMember() be improved to match the call back functionality of 
AgentCallBackLogin()?


AgentCallBackLogin() is deprecated beginning with 1.4. You can use
AddQueueMember() in combination with the Local/ channel to do what
you're looking to do above.

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Re: [asterisk-users] extensions.conf strangeness

[Brian Candler]

On Fri, Sep 29, 2006 at 07:49:00PM +0200, Michael Neuhauser wrote:
> The order of include statements is important in 1.2, I don't know if
> this still holds for trunk/1.4. Could you please try to include the
> 'invalid' context as the last one (i.e., AFTER "include => test", not
> before) in both internal and from-sip and then test again?

Yes, this works - both contexts now behave the same.

But what I don't understand is, why it worked in one context but not in the
other, when both just included the same four other contexts in the same
order. Is the context merging non-deterministic? Or is it somehow sensitive
to whether the incoming call came from zaptel or SIP?

Regards,

Brian.
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Re: [asterisk-users] WAS: 64 analog phones NOW: Selection criteria and recipie for a good Asterisk install [long]

[Tzafrir Cohen]

On Thu, Sep 28, 2006 at 03:17:15PM -0400, Jeronimo Romero wrote:
> Has anyone tried RedFone?? It is supposed to offload a lot of that bus
> overhead to the external unit doing TDMoE. 

Offloading? What exactly?

A quad E1 (4 E1 cards, much more than is needed for the 64 lines
mentioned in the topic) takes 8Mb of actual data per second, overheade
not included.

It doesn't take much horsepower: A decent single-CPU P4 server could
easily handle the full capacity of that card, assuming that there are no
other load factors such as conferences, compressions and such.


However if you get a network segment into the picture, you now have to
take into account not only your server and the other server, but also
all the network infrastructure in between. E.g.: the switch(es), QoS,
etc.

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
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RE: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN

[Wolfgang_Borgon]

David,
Yes, I've also forwarded port 4569 to the server. 
Since the router is forwarding to the server, I cannot
forward it to the client as well -- however, as the
client isn't going out past the LAN, it shouldn't
matter... unless there's something else going on that
I don't know about.
Thanks
Wolfgang

--- David J Carter <[EMAIL PROTECTED]> wrote:

> Wolfgang wrote:  -
> 
> I've already sunk several hours into this without
> any
> real progress, so I'd really appreciate any help  My
> task is simple -- establish a connection between a
> softphone on XP ProSP2 to a Asterisk server on Linux
> FC4 over a LAN through a Netgear router. The server
> will then go out to a PSTN termination service.
> 
> Thus far, the PSTN termination connection works fine
> -- I've opened up 4569 with iptables, and forwarded
> 4569 to the server IP.  I am not, however, having
> any
> luck connecting the softphone to the server.
> 
> I can telnet, ftp, and http to the server, but not
> IAX2. Iaxping times out, registration by Idefisk and
> Firefly also times out.  
> 
> The server fails to see the client as well.  
> 
> Here's a portion of my iax.conf:
> 
> [client]
> type=friend
> username=client
> secret=**
> host=192.168.1.40
> context=clientcon
> 
> and extensions.conf:
> 
> [clientcon]
> exten => 2278,1,Dial(IAX2/client)
> 
> 
>
==
> You say you have 4569 configured in iptables, what
> about the netgear router?
> 
> Have you port forwarded 4569 there?
> 
> Dave
> 
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Re: [asterisk-users] real time billing system

[Pato Valarezo]

Chapeti wrote:
Hola pato, hasta donde yo sé no hay nada que sea código abierto, lo que 
me parece mas fácil es que te hagas
uno propio, echale una ojeada a lo que hay en 
http://www.voip-info.org/wiki-Asterisk+manager+API, lo
único que haría falta sería un poco de conocimientos deVB 6 y de como 
trabajar con sockets ( cosa que no es nada del otro mundo ).


Saludos.


mmm... bueno no estaba buscando precisamente algo que sea abierto, 
simplemente algo que me ayude a instalar un pequeño locutorio con 
telefonos sip y con 4 salidas zap.
Para lo que me comentas del AMI, muy interesante, la verdad que se 
pueden hacer maravillas... aunque no lo hiciera en VB, mas bien en algo 
mejor como python!. Voy a buscar que encuentro y si no hay nada 
adaptable me pondré manos a la obra con esto.


gracias por la información.

saludos

--
patoVala
Linux User#280504
Hablando en http://www.elprimoalcahuete.com
"El tiempo cura los dolores y las querellas porque cambiamos. Ya no 
somos la misma persona. -- Blaise Pascal. (1600-1662) Filósofo y 
escritor francés. "

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Re: [asterisk-users] Problems with DISA

[Shidan]

Well this is how I got around it, and since it's just a test box it'll
do. I commented out most of  the function that's responsible for
playing the tones so that now it is just a wrapper around this:
"ast_tonepair_start(chan, 350, 440, 0, 0);" So looks like somethings
off with the data coming back from indications.conf ( but the file is
correct).

So if it's found to be a more general problem thats something to look at.

Thanks for all your responses.

---
Shidan Gouran

On 9/29/06, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:

Make sure you have a /etc/asterisk/indications.conf

Shidan wrote:
> For some reason I'm having problems with DISA. This is what I have:
>
> exten => s,1,Answer()
> exten => s,2,DigitTimeout(5)
> exten => s,3,ResponseTimeout(10)
> exten => s,4,DISA(no-password|from-internal)
>
> I can generate tones with no problems on this system
> But I cant hear a dialtone using DISA, anyone experienced
> this problem? Also, I'm running this in the cusom confs for
> freepbx.
>
> Just to save time, I'm not looking for a solution that doesn't
> use the Disa app.
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RE: [asterisk-users] Queue AddQueueMember()

[Douglas Garstang]

> -Original Message-
> From: BJ Weschke [mailto:[EMAIL PROTECTED]
> Sent: Thursday, September 28, 2006 2:26 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Queue AddQueueMember()
> 
> 
> On 9/28/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:
> > All,
> >
> > I've recently been told that the AgentCallBacklogin() 
> application is buggy, and I should not use it. Apparently I 
> should use AddQueueMember() instead. I see though that 
> AddQueueMember() does not take the location to call back as 
> an argument.
> >
> > We have remote agents that are available via PSTN access 
> only. With AgentCallBackLogin() they can enter their PSTN 
> phone number, and Asterisk will call them back at that number 
> when they get a queue call. Can AddQueueMember() do that?
> >
> > Is AgentCallBackLogin() going to be deprecated at some 
> point? Will AddQueueMember() be improved to match the call 
> back functionality of AgentCallBackLogin()?
> 
>  AgentCallBackLogin() is deprecated beginning with 1.4. You can use
> AddQueueMember() in combination with the Local/ channel to do what
> you're looking to do above.

The queue function AgentCallBackLogin() would take the name of an agent as an 
argument, and would log that agent into the queue they where associated with. 
We programmed our appearances on our Polycom phones to have a different 
appearance for each queue, and we'd send the caller id of the appearance as the 
name of the agent, thereby removing the need for the agent to enter their agent 
number. 

I see though that the AddQueueMember() function requires a queue name as an 
argument. That makes the dialplan logic more complex as we somehow have to 
magically send the queue name that this agent belongs to from the phone to the 
dialplan.

Doug.
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[asterisk-users] 480i phone: Is there a trick to registering with * ??

[Colin Anderson]

Running * 1.0.9, sip.conf allow=all is set.

Based on the advice of -users earlier this week I've ordered an Asstra 480i
CT for evaluation. Phone is up, sees Asterisk, tries to register, Asterisk
refuses. I though it might be codec mismatch so I specified allow=all. Valid
account, password OK, codec OK, just Asterisk rejects the registration
attempt. Here is what comes out of the syslog of the phone:

09-29-2006  16:22:54Local7.Debug192.168.1.223
<013><010><<=IN=192.168.1.223: Received SIP packet from:
192.168.1.46:5060<013><010>SIP/2.0 401 Unauthorized<013><010>Via:
SIP/2.0/UDP 192.168.1.223;branch=z9hG4bKcbf13ba80<013><010>From: 7028247
;tag=3727ada04d20136<013><010>To: 7028247
;tag=as39cd5209<013><010>Call-ID:
[EMAIL PROTECTED]<013><010>CSeq: 26925325
REGISTER<013><010>User-Agent: Asterisk PBX<013><010>Allow: INVITE, ACK,
CANCEL, OPTIONS, BYE, REFER<013><010>Contact:
<013><010>WWW-Authenticate: Digest realm="asterisk",
nonce="64a24b82"<013><010>Content-Length:
0<013><010><013><010><<=IN=END SIP packet

And Asterisk sip debug:


Sip read:
REGISTER sip:192.168.1.46:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.223;branch=z9hG4bKcbf13ba80
Max-Forwards: 70
Content-Length: 0
To: 7028247 
From: 7028247 ;tag=3727ada04d20136
Call-ID: [EMAIL PROTECTED]
CSeq: 1580356212 REGISTER
Contact: 7028247 ;expires=1
Allow-Events: talk,hold,conference
Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO
User-Agent: Aastra 480i Cordless/1.3.1.1095 Brcm Callctrl/1.5 MxSF/v3.2.6.26


12 headers, 0 lines
Using latest request as basis request
Sending to 192.168.1.223 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.223;branch=z9hG4bKcbf13ba80
From: 7028247 ;tag=3727ada04d20136
To: 7028247 ;tag=as1c8c1d8b
Call-ID: [EMAIL PROTECTED]
CSeq: 1580356212 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0

?? help anyone ??
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[asterisk-users] FW: 480i phone: Is there a trick to registering with * ?? <--forg ot my sip.conf

[Colin Anderson]

[8247]
username=8247
type=peer
secret=
quaify=no
port=5060
pickupgroup=
nat=never
[EMAIL PROTECTED]
host=dynamic
dtmfmode=rfc2833
allow=ulaw
allow=alaw
context=from-internal
callerid="Colin Anderson" <702-8247>
canreinvite=no

Tried "peer" , "friend" "allow=all" etc no go
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Re: [asterisk-users] DSL router with integrated SIP proxy?

[Jerry Jones]

We are trying a couple of the Intertex - seems to work so far


On Sep 29, 2006, at 2:59 PM, Andrew Joakimsen wrote:


The VoIP version of DD_WRT runs Ser by default

On 9/24/06, David Gagnon <[EMAIL PROTECTED]> wrote:

You could take a WRTSL54gs, install openwrt then openser

David

-Message d'origine-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de  
Steve Kennedy

Envoyé: 24 septembre 2006 08:47
À: asterisk-users@lists.digium.com
Objet: Re: [asterisk-users] DSL router with integrated SIP proxy?

On Thu, Sep 21, 2006 at 10:11:43PM +0100, Brian Candler wrote:

> Does anyone here know of an ADSL router with integrated SIP proxy?

Netscreen 5GT ADSL, it has what's called an ALG (application layer
gateway) and it does indeed support SIP. Full featured firewall  
etc too.



Steve

p.s Hi Brian :)

--
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com
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Re: [asterisk-users] real time billing system

[Guillermo Salas M.]

On Fri, 2006-09-29 at 16:48 -0500, Pato Valarezo wrote:
> Chapeti wrote:
> > Hola pato, hasta donde yo sé no hay nada que sea código abierto, lo que 
> > me parece mas fácil es que te hagas
> > uno propio, echale una ojeada a lo que hay en 
> > http://www.voip-info.org/wiki-Asterisk+manager+API, lo
> > único que haría falta sería un poco de conocimientos deVB 6 y de como 
> > trabajar con sockets ( cosa que no es nada del otro mundo ).
> > 
> > Saludos.
> 
> mmm... bueno no estaba buscando precisamente algo que sea abierto, 
> simplemente algo que me ayude a instalar un pequeño locutorio con 
> telefonos sip y con 4 salidas zap.
> Para lo que me comentas del AMI, muy interesante, la verdad que se 
> pueden hacer maravillas... aunque no lo hiciera en VB, mas bien en algo 
> mejor como python!. Voy a buscar que encuentro y si no hay nada 
> adaptable me pondré manos a la obra con esto.
> 

Que tal Pato :)

I'm using a2billing at my cybercafe and works very well.

You can use starshop-oss as well the setup instructions are at: 
http://www.starshop-online.com/howto/how_to_setup_starshop.htm

I preffer a2billing because is giving me more features like having two
or more providers for the same destination and LCR.

Saludos, 

> gracias por la información.
> 
> saludos
> 
-- 
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular  : +593 9 985 5138
e-mail   : [EMAIL PROTECTED]
www  : http://www.manta.telconet.net
   http://www.telcocarrier.net

Linux User: 255902

Beat me, whip me, make me use Windows!

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

Please avoid the Top Posting, see
http://es.wikipedia.org/wiki/Top-posting

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Re: [asterisk-users] pstn failback

[stan ford]

Lacy,  can you confirm what i was saying about SIP Phones. if i fail from my voip connection to my pri, would i need to swap out my SIP phones with another type of digital phone?Lacy Moore - Aspendora <[EMAIL PROTECTED]> wrote:   a couple things, if you guys could clear up for me.     A) If i have a bunch of sip ip phones, and if i were to fail to my PRI. I should still be able to use my ip phones right? i assume the signal will be changed by my T1 card? and the reverse i would assume is true for incoming calls.      Outgoing could be handled by the dial plan.  Incoming would have to be something worked
 out with the providers.   B) How about DID's, how would that be handled. is there a DID failover as well? I have my VOIP service with one company, if i had my PRI service with another. how would those DID's get failed to the other provider, if thats even possible at all in a timely manner.      Not sure how this would be handled.     C) also are failover pri's generally cheaper that their active counterparts?     thanks alot.   ___--Bandwidth and Colocation provided by
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Re: [asterisk-users] Extension Numbering

[Norbert Zawodsky]

And Hi again,

I wrote:
> I think I can remember something from the "Asterisk-TFOT" book saying
> that one must not use 1-digit extension numbers. But I can't remember
> that very well and can't find it in the book any more
>   
I found it! On page 90 the book says:


... (Well, almost. Extensions must be shorter than 80 characters long,
and you shouldn’t use single-character extensions for your own
use, as they’re reserved.) ...


O.k. - This answers my first question ("if there is disadvantage if we
use only 1 digit extensions").
But what for are single-character extensions "reserved" ?

Thanks,
Norbert

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Re: [asterisk-users] pstn failback

[Lacy Moore - Aspendora]

If you are using Asterisk, I wouldn't think so.  Asterisk will choose what media the outbound call is on.  If no Voip connections are available, it can continue (by what you have defined in the dial plan) on the a pri.

On 9/29/06, stan ford <[EMAIL PROTECTED]> wrote:

Lacy,
can you confirm what i was saying about SIP Phones. if i fail from my voip connection to my pri, would i need to swap out my SIP phones with another type of digital phone? 
Lacy Moore - Aspendora <[EMAIL PROTECTED]>
 wrote:




 
a couple things, if you guys could clear up for me.
 
A) If i have a bunch of sip ip phones, and if i were to fail to my PRI. I should still be able to use my ip phones right? i assume the signal will be changed by my T1 card? and the reverse i would assume is true for incoming calls. 

 
Outgoing could be handled by the dial plan.  Incoming would have to be something worked out with the providers.

 
B) How about DID's, how would that be handled. is there a DID failover as well? I have my VOIP service with one company, if i had my PRI service with another. how would those DID's get failed to the other provider, if thats even possible at all in a timely manner. 

 
Not sure how this would be handled.  

 
C) also are failover pri's generally cheaper that their active counterparts?
 
thanks alot. 
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http://lists.digium.com/mailman/listinfo/asterisk-users-- Lacy MooreI'm the guy that doesn't give a damn about anyone's problems but my own... 
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Re: [asterisk-users] Extension Numbering

[Lacy Moore - Aspendora]

I'm not too sure on that.  I don't really know, unless maybe for an IVR and you want a menu of options (which can only be 1 digit) and also dial direct to the extension.  That would limit you in that regard.
 
I have a suggestion regarding dial plan.  When I first started I saw no reason to have to dial 9 first for outside calls.  Because I wanted to be able to dial out from the missed calls list, I chose to eliminate the dial 9 requirement.  I'm now regretting it, primarily because 711 is a valid number, but also a parking spot.  I can handle 911, and I can handle 411, but 711 is going to be a pain.

 
Just wanted to give you something to consider when you are starting out.  Something I didn't consider. 
On 9/29/06, Norbert Zawodsky <[EMAIL PROTECTED]> wrote:
And Hi again,I wrote:> I think I can remember something from the "Asterisk-TFOT" book saying
> that one must not use 1-digit extension numbers. But I can't remember> that very well and can't find it in the book any more>I found it! On page 90 the book says:... (Well, almost. Extensions must be shorter than 80 characters long,
and you shouldn't use single-character extensions for your ownuse, as they're reserved.) ...O.k. - This answers my first question ("if there is disadvantage if weuse only 1 digit extensions").
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Re: [asterisk-users] Extension Numbering

[Michiel van Baak]

On 01:35, Sat 30 Sep 06, Norbert Zawodsky wrote:
> And Hi again,
> 
> I wrote:
> > I think I can remember something from the "Asterisk-TFOT" book saying
> > that one must not use 1-digit extension numbers. But I can't remember
> > that very well and can't find it in the book any more
> >   
> I found it! On page 90 the book says:
> 
> 
> ... (Well, almost. Extensions must be shorter than 80 characters long,
> and you shouldn?t use single-character extensions for your own
> use, as they?re reserved.) ...
> 
> 
> O.k. - This answers my first question ("if there is disadvantage if we
> use only 1 digit extensions").
> But what for are single-character extensions "reserved" ?

There's:
a (*)
s (start)
t (timeout)
i (invalid)
h (hangup)
And maybe others, those come to mind right this second since
I use them a lot. They want to be sure they have 21 more
possible default available extensions.
That's why it's adviced to not use them to prevent trouble
with newer versions where your single-character extensions
clashes with some default builtin one.

For readability it's also better to use descriptive
extensions instead of singe character ones.

-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

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

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Re: [asterisk-users] Sip answer one side , ring other side

[Leo Ann Boon]

antonio wrote:

Hi,
the scheme is this :
 
xlite  ---> Asterisk ---> SIP gateway  ---> PSTN
 
 
When i make a call with xlite (sip) to asterisk on the display of 
xlite i see that the call is connected but the phone is still ringing ..
You must configure your gateway to NOT answer the call before making the 
PSTN call. Some gateway call that '1-step' dialing.


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[asterisk-users] SIP phones not talking

[joe, at j4computers]

Setting up a new system, have two sip phones that give dial tone and appear to 
dial, but do not complete, 
giving a busy.

Watching the CLI thing, get this message,

-- Goto (macro-record-enable,s,4)
-- Executing AGI("SIP/200-0825b648", 
"recordingcheck|20060929-195420|1159574059.5") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
tel1*CLI> Failed to execute '/var/lib/asterisk/agi-bin/recordingcheck': No such 
file or directory


Yet, the same thing seems to have executed prior, without error.

Any suggestions?

joe
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[asterisk-users] Fax detection ...

[phil . dawson]

Hi All,

I'm having trouble detecting faxes reliably.  I'm using one analog line for
both voice and fax.  Sometimes this works but a lot of the time it doesn't
and I was wondering if anyone knew why.  This is how I imagined it to work:

Caller calls in and as there is no fax tone asterisk processes the call as
a voice call.
Fax machine calls in, asterisk detects this and sends the call to the fax
extension.

I have eliminated my hylafax from the problems I'm experiencing as the call
isn't getting that far.

My configuration:

pstn ( analog ) --- > Asterisk ( TDM 400P )  --- > Linksys PAP2T-UK  --- >
Modem  --- > Hylafax


Something else I've recently come across is a lot of fax machines wait for
the callee to present a fax tone before it will communicate.  How would I
present a tone from asterisk when someone calls to be able to detect fax or
voice or is this not possible?



extensions.conf

exten => s,1,Answer
exten => s,n,LookupCIDName
exten => s,n,Wait(3)  ; <-- wait to see if fax  (
should I need this )
exten => s,n,Goto(mainmenu,7,1) ; < -- if not fax process
ivr


exten => fax,1,Dial(${PHONES4},40,Ttr)  ; <-- dial our ata with
connected fax machine
exten => fax,n,Hangup()


zapata.conf

faxdetect=incomming



Thanks in advance

Phil

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[asterisk-users] Re: max number of devices in hint

[Lacy Moore - Aspendora]

In case anyone runs into this, I'm posting what I ended up doing...
 
I tracked down a file under include/asterisk/channel.h
 
This file defines the length of the variables in this section:
 
/*! Max length of an extension */#define AST_MAX_EXTENSION 80#define AST_MAX_CONTEXT 80#define AST_CHANNEL_NAME 80 
Not knowing which one, I changed all three to 255, recompiled, and my hints work now.  If I were to guess, I would guess that AST_CHANNEL_NAME is the variable that needs to be changed.  At this point, I've got a relatively small setup with 2 gigs of ram and over 1 gig free after compiling, so I don't think I'm taking that big of a hit.

 
 
On 9/26/06, Lacy Moore - Aspendora <[EMAIL PROTECTED]> wrote:

I have one extension that rings in many places.  It has just come to my attention that I can only monitor 4 devices within a hint.
 
Ex:
 
exten => 132,hint,SIP/DEVA&SIP/DEVB&SIP/DEVC&SIP/DEVD
 
if I add SIP/DEVF, DEVF is not monitored.
 
Is anyone else monitoring more than 4 devices, and if so, what version are you running?
 
I'm running 1.2.12.1.  I thought at first this may be a phone issue, but by running SHOW HINTS on the CLI, it shows InUse using the first four devices.  The fifth device shows IDLE when it is, in fact, in use.  I also switched places for DEVD and DEVE thinking maybe it was something with DEVE.  When I switch places, DEVE shows as expected and DEVD does not show. 

 
Any ideas?
-- Lacy Moore -- Lacy MooreI'm the guy that doesn't give a damn about anyone's problems but my own... 
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Re: [asterisk-users] 7940 vs. 7941

[Greg Oliver]

On Thu, 2006-09-28 at 07:54 -0500, Tom wrote:
> At 05:39 AM 9/28/2006, you wrote:
> >Any pros / cons on getting one over the other ? I was wondering what 
> >the main differences were.
> 
> New phones (7941) support 802.3af POE.  Old phones only Cisco special 
> POE.  New phones don't work with old SIP images.  Only new unified 
> SIP/SCCP images.
> 
> New phones have a higher resolution display.  New phones have some 
> lighted buttons.
> 
> Tom
> 

The new phones also run Java for their OS, so they are quite a bit
slower than the 40/60 series for menus, etc...

Their graphical displays are much higher resolution then the older
models as well.

-Greg

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