Re: [Asterisk-Users] Soekris and Asterisk

[Christopher Dobbs]

trixter http://www.0xdecafbad.com wrote:

  On Wed, 2005-10-12 at 17:46 -0700, Paul Mahler wrote:
  
  
You need about 30MHz per channel. That means the Soekris can only handle part
of a T1, it will never handle a quad span. 

Paul


  
  
How was that determined?  

I have a problem with a plain number like that, which may have been
taken into account, why I am asking...  

Different cpus operate differently, taking more or less time to complete
certain functions.  Instruction optimization can go a long way if those
instructions are used (not terribly likely if its just pushing bits but
there are some for just that).

Additionally there is no codec processing (presumably) with TDMoE, does
the 30MHz take into account any codec processing or is it literally
30MHz (on what cpu class?!) for just pushing bits?

There are other factors, but you did say 'about' so they are optional to
this conversation, ie other IRQs on the box, potential for device
polling, etc.  A tuned system for that specific task (pushing bits
between a TDM card and ethernet via TDMoE) may be able to operate at a
lower clock speed per channel, but that isnt as important for the
initial questions.



  
  

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MHZ is not a valid way of gauging performance.  It's all about the MIPS
(Millions of Instrictions Per Second), Baby :).

I was testing with some of the Soekris boards about a year ago for an
client, the need was to make a TDMoE -> TDMoE router for a wireless
network. (Yes I know that that is a stupid idea, and I told the client
that it was a waist of his money to have me try.) the board I was using
I think was the 4801, not sure thoe (It was a year ago) but it would
pust 48 TDMoE channels at once over 100BaseT ok.  So I would think that
It would.  I was using a customized linux distro, (as in one I created)
contact me off list if you would like a copy of the distro.

--
Christopher Dobbs
Wireless Administrator
Valario Inovations



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RE: [Asterisk-Users] supermicro with asterisk and tdm cards

[Anton Krall]

I guess the 2U is not bad... Im going to call supermicro and check what they
have. What kind of CPU are you using guys? Seems supermicro has everythiung
except the CPU and the HD right? 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Kevin Bockman
|Sent: Wednesday, October 12, 2005 3:05 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] supermicro with asterisk and tdm cards
|
|Cory Andrews wrote:
|> Yeah I should have picked up on that, single PCI Riser in 
|this one, so 
|> 1 card.  I don't know of any 1U solution out there that 
|would give you 
|> 3 PCI slots to work with, I think you'll have to go to a 2U at least 
|> to achieve this.
|I saw the Dell PowerEdge 1850 has 2 PCI-X on separate busses.  
|That's the only one I've ever seen.
|
|
|Kevin
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[Asterisk-Users] Perplexed - IAX trunk == jitterbuffer or no jitterbuffer

[Jason Walker]

I have 4 * servers interconnected with IAX trunks. Three are on a local LAN,
one is accessible over a VPN tunnel out of the office. The IAX peer status
(iax2 show peers from the CLI) will sometimes show upwards of 300ms.
Considering the lag and distance, I am not entirely surprised.

Anyway - my question falls towards the jitterbuffer settings in the
iax.conf. 

Should I or should I not? I seem to come across one document that says to do
it to only find another document that says this is not the best option for
my particular installation. So I am now perplexed.

I did updated the MAX_TIMESTAMP_SKEW value in rtp.c to an increased value
(found that in one of the bug trackers) and then recompile. But the other
settings, let alone to use the jitterbuffer at all, is still a quandary.

These are the latest values I am using:

jitterbuffer=yes
dropcount=2
maxjitterbuffer=200
maxexcessbuffer=40
minexcessbuffer=5
jittershrinkrate=1

I have changed bandwidth and tos to maximize bandwidth and reliability. What
I end up with are calls that sound like the far end is in a helicopter. I
can only assume that the packets are ending up out of order. Or...?

Any help, assistance, guidance, and past experience is GREATLY appreciated!

Thanks!

Jason

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Re: [Asterisk-Users] Integrated T1

[Mitchel Constantin]

Yes it will support it, you should look up "HDLC" on the wiki...I went
through this a year ago and had a hard time setting it up. It might be
easier now though. I would recommend going another route and getting
the data brought in seperately with it's own router. You'll also have
better redundancy that way.

Good luck,
Mitchel

On 10/12/05, Samy Antoun <[EMAIL PROTECTED]> wrote:
> Hi,
>
> We have a Data/Voice service supplied through an
> integrated T1.
> Does anyone know if Digium T1 card will support the
> splitting of the Voice and Data?
>
> Regards.
>
>
>
>
>
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Re: [Asterisk-Users] displaying a message on the Snom 320 using sipsak

[Sven Fischer (support)]

Hi,

for the snom360 it is working the same way. Use firmware version 4.3 and be 
aware that the message is send to a specific SIP line and the phone is 
displaying it only if this SIP line is the current active line (outgoing 
identity) symbolised by a black phone (snom360) or the text in brackets 
(snom320).

Regards,

Sven Fischer

On Wednesday 12 October 2005 20:20, Franklin Webb wrote:
> Greetings fellow list members,
> It seems like a lot of people have been having trouble getting
> indicators working on the Snom phones, myself included.  Recently I was
> able to get the "desktop" functionality of sipsak to work on my Snom320,
> and I thought I would share what I could with the list.  For those not
> familiar this will replace the standard display when you are not on a
> call (normally showing the registered extension) with a text message of
> your choosing.  Our intent is to update this when our agents log into,
> and out of, queues.  This will give a visual indicator for agents and
> supervisors in our call center as to whether or not the phone is logged
> in, which is a large concern for us, and probably any call center.
>
> For the record I tried this with a Snom360 also and could not get it
> working.
>
> 1.  Setup the phone in Asterisk as normal
> 2.  Get and install sipsak.  It can be found at http://sipsak.org/
>   (can be on any machine on your network, we used a
> Fedora Core 3 machine for this).
> 3.  In the Snom320 Configuration, under the "SIP" tab of your extensions
> line (Line 1 for me) make sure "Support Broken Registrar" is set to "on"
> 4.  In the Snom320 Configuration, under "Advanced" make sure "Filter
> Packets from Registrar" is set to "off"
> 5.  In the Snom320 Configuration, under "Advanced" under "Network
> identity (port):" set it to "5060" (you might be able to use a different
> port in here and in the sipsak command, but this is what worked for me.
> 6. Reboot the phone (just to be sure the settings take)
>
> Then use the following sipsak command:
>
> sipsak -vvv -M -O desktop -B "Test Msg" -r 5060 -s
> sip:[EMAIL PROTECTED]
>
> where:
> "Test Msg" is the message you want displayed.  To turn the message
> off just set it to empty string ("").
> 5060 is the port, you could try another port here if you set your
> phone to another port under "Advanced"
> 6670 is the extension of the phone
> 192.168.51.251 is the IP of the PHONE, not the Asterisk server.  It
> does not appear that you can use the IP of the Asterisk server.
>
> You can get a list of phones with IPs using the Asterisk command "sip
> show peers".  Our intent is to build a simple database matching
> extension to IP and then execute sipsak commands from a script, probably
> in the manager API, when agents log in and out that will update the
> phone display accordingly.
>
> I hope this is helpful to some of you.
>
> Franklin Webb
> InterMedia Marketing Solutions

-- 
---
See our FAQs at: http://www.snom.com/faq0.html?&L=1
Whitepapers at:  http://www.snom.com/white_papers.html
---
snom technology AG   Gradestraße 46 D-12347 Berlin
Sven Fischer fax +49 30 39833111
mailto:[EMAIL PROTECTED]   http://www.snom.com
---
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Re: [Asterisk-Users] Broadvoice Outages?

[Samy Antoun]

--- Nate Kapi <[EMAIL PROTECTED]> wrote:
> I've been having a lot of problems with Broadvoice
> lately. Anyone else
> been without service for extended periods of time
> this week?

Service is down right now 




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[Asterisk-Users] Broadvoice Outages?

[Nate Kapi]

I've been having a lot of problems with Broadvoice lately. Anyone else
been without service for extended periods of time this week?
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Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?

[Craig Guy]

I have downloaded iaxmodem and gone through the readme but not yet installed 
it.  I currently use rxfax to receive in the vicinity of 1200 faxes per day 
and 5000 or more pages (faxes vary from single page to 30 pages) per E1, 
with a peak load of about 12 concurrent inbound faxes to rxfax.  Best I can 
tell my failure rate is about 0.8%.  I have been testing using Hylafax for 
faxout with an 8 port analog fax modem card and a couple PAP2NA's and this 
works well, but I am very much looking forward to checking out iaxmodem. 
Especially if using Hylafax will give me ECM.


Craig

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


Sent: Thursday, October 13, 2005 10:47 AM
Subject: Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?



Darren Nickerson wrote:

We prefer the Eicon Diva server and Brooktrout TR1034 boards, which are 
known to work well with HylaFAX since we've had our share of headaches 
with the 2977's.



Well, part of my preference for the 2977s involves my strong dislike for 
the way that the Diva Servers and BrookTrouts do things.  It's enough of a 
dislike to get me over the learning curve of how to properly set up the 
2977s for HylaFAX use.


Having said that, I'm excited to see Lee and Steve improving IAXmodem and 
the underlying SpanDSP library, and look forward to the day that is 
performs similarly (or better) to the DSP-laden boards we presently 
favor!



If your favor involves V.34 then it may be a while before the relevant 
patents expire.


Lee.

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[Asterisk-Users] Integrated T1

[Samy Antoun]

Hi,

We have a Data/Voice service supplied through an
integrated T1.
Does anyone know if Digium T1 card will support the
splitting of the Voice and Data?

Regards.





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[Asterisk-Users] SIP to SIP no audio help

[Michael Furdyk]

Hi everyone!

I've been working on setting up an Asterisk server and my two Digium
cards I ordered will arrive tommorow, so I'm excited to plug some 'real'
old-school lines into it.

But tonight I've been testing with some of our staff around the world,
and while handing off 'real' (PSTN - over VoIP using Voicepulse) calls
to SIP and SIP calls to VoIP PSTN works fine, SIP to SIP calls provide
no audio, and just this message on console:

-- Executing NoOp("SIP/322-3edd", ""call for "331") in new stack
-- Executing Dial("SIP/322-3edd", "SIP/331|60|tr") in new stack
-- Called 331
-- SIP/331-fd48 is ringing
-- SIP/331-fd48 answered SIP/322-3edd
-- Attempting native bridge of SIP/322-3edd and SIP/331-fd48

My sip.conf has nat on and canreinvite=no, and those were the only
suggestions I could find. Help would be greatly appreciated! We are
really excited about the potential of Asterisk.

Sip.conf:

[322]
type=friend
context=softphone ; match with the outgoing context in extensions.conf
host=dynamic ; This device needs to register callerid="Michael Furdyk"
 nat=yes ; X-Lite is behind a NAT router canreinvite=no ;
Typically set to NO if behind NAT allow=all ; codec choice: GSM consumes
far less bandwidth than ulaw
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info



Cheers,

- Michael

Michael Furdyk
Director of Technology, TakingITGlobal
http://www.takingitglobal.org/
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Re: R: [Asterisk-Users] Problem setting up TDM22B card

[Somesh S Shanbhag]

modprobe zaptel is successful. When I do lsmod zaptel
is loaded.

Regards,
Somesh S. Shanbhag

--- Lyle Giese <[EMAIL PROTECTED]> wrote:

> I have not seen the output of modprob zaptel in this
> thread, which has 
> to take place before loading the other kernel
> drivers.
> 
> Lyle
> 
> 
> so
> mesh s wrote:
> 
> >Hi,
> >
> >I changed the mother board (MB) but it is giving
> still
> >the same problem.
> >  
> >
> >ouput of dmesg|tail 
> >  
> >
> >f6 != 58
> >f7 != 59
> >f8 != 58
> >f9 != 59
> >fa != 58
> >fb != 59
> >fc != 58
> >fd != 59
> >fe != 58
> >Freshmaker failed register test
> >  
> >
> >and I have also configured zaptel.conf correctly.
> >
> >Whatz next? Can I assume that it is a hardware
> >problem?
> >
> >Regards,
> >Somesh S. Shanbhag
> >
> >
> >--- John Novack <[EMAIL PROTECTED]>
> wrote:
> >
> >  
> >
> >>somesh s wrote:
> >>
> >>
> >>
> >>>Hi,
> >>>
> >>>I didn't get any solution in the mailing list.
> >>>  
> >>>
>
>>[http://asterisk.linkx.net/asteriskusers/200409/msg01167]
> >>
> >>
> >>>What should be the next step?
> >>>
> >>>Changing the machine???
> >>>Is it machine dependent?...
> >>>
> >>>Regards,
> >>>Somesh S. Shanbhag
> >>>
> >>> 
> >>>
> >>>  
> >>>
> >>Have you talked with Digium support?
> >>
> >>Their answer almost always is:
> >>
> >>"Try another Motherboard"
> >>They won't supply a list that is known to work,
> only
> >>ones that are known 
> >>NOT to work.
> >> From my limited experience, even if the MB says
> it
> >>is PCI 2.2, the TDM 
> >>card may or may not work.
> >>
> >>If you don't want to change machines, then  use an
> >>ATA or two Sipura's 
> >>work great.
> >>
> >>John Novack
> >>
> >>
> >>
> 
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Re: [Asterisk-Users] How can I use different languages (Chinese, Cantoneese)?

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

On Thu, 2005-10-13 at 12:17 +0800, Ronald Wiplinger wrote:
> I want to give the users the announcements as they subscribed to. The 
> announcements should be in English, Chinese, Cantonese, according to 
> their phone number. How can I do that? I can hardly make for each number 
> a different context!!!
> 

http://www.voip-info.org/wiki-Asterisk+cmd+SetLanguage

either use an agi to fetch the settings, or dbget()

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


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[Asterisk-Users] SIP to SIP no audio help

[Michael Furdyk]

Hi everyone!

I've been working on setting up an Asterisk server and my two Digium
cards I ordered will arrive tommorow, so I'm excited to plug some 'real'
old-school lines into it.

But tonight I've been testing with some of our staff around the world,
and while handing off 'real' (PSTN - over VoIP using Voicepulse) calls
to SIP and SIP calls to VoIP PSTN works fine, SIP to SIP calls provide
no audio, and just this message on console:

-- Executing NoOp("SIP/322-3edd", ""call for "331") in new stack
-- Executing Dial("SIP/322-3edd", "SIP/331|60|tr") in new stack
-- Called 331
-- SIP/331-fd48 is ringing
-- SIP/331-fd48 answered SIP/322-3edd
-- Attempting native bridge of SIP/322-3edd and SIP/331-fd48

My sip.conf has nat on and canreinvite=no, and those were the only
suggestions I could find. Help would be greatly appreciated! We are
really excited about the potential of Asterisk.

Sip.conf:

[322]
type=friend
context=softphone ; match with the outgoing context in extensions.conf
host=dynamic ; This device needs to register
callerid="Michael Furdyk" 
nat=yes ; X-Lite is behind a NAT router
canreinvite=no ; Typically set to NO if behind NAT
allow=all ; codec choice: GSM consumes far less bandwidth than ulaw
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
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Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?

[Lee Howard]

Darren Nickerson wrote:


I'm curious though, in an earlier message you wrote:

"Sending is also quite  good, however, there are some quirks with the 
Patton

firmware which need to be resolved for me to be completely delighted."

Are these issues you refer to just cosmetic 'nice to haves' or do they 
actually impact the outbound success rate? Is the 2977's outbound in 
Class 1 as solid as the Mainpine or Multitech boards now? Our 
experience has shown that is not the case ...



Having worked extensively with modems such as MultiTechs and MainPines 
which use the Lucent/Agere Venus chipset, modems such as Comtrol 
RocketModems (II not I or III) that use the Rockwell/Conexant K56 
chipset, and the Patton 2977 (which along with the Digi AccelePort RAS 
uses an ADI chipset) I can say unequivocally that the former two _can_ 
currently outperform the latter.


But allow me to put that performance difference into perspective ... 
depending upon the locale I can see 99.9% success rates (ratio is pages 
without error divided by total pages received) with MultiTechs, 
Mainpines, and Comtrols.  By comparison, the performance that I see with 
the Patton 2977 averages somewhere between 99.7% and 99.8% success 
ratio.  That said, I have seen deployments of the former two chipsets 
underperform those statistics under "normal" conditions (say 98% to 99%) 
simply because they frequently send to or receive from a very 
problematic fax machine on the other end... and still the owners of 
those fax servers are quite happy with how those perform.  In fact, I 
know of HylaFAX deployments that use USR modems and suffer at around 95% 
success ratio, and the problems even there aren't enough to motivate 
them to buy better modems.  So the performance of the Patton 2977 
shouldn't be considered "bad"... just not as good as I think it should 
be.  Furthermore, when we're talking about digital phone lines and 
digital fax equipment, I should think that could eliminate 50% of the 
potential pitfalls right there... and if the other end is also using 
digtial then communication should be flawless, right?  Well, that's my 
idea anyway... it doesn't work out that way.


I'm a finicky faxer.  One error in a thousand pages may seem 
extraordinary to some, but when thousands upon thousands of pages are 
handled every day the 0.1% is annoying if not frustrating.


I could elaborate on the specific flaws in the Patton cards, I suppose, 
but this isn't the time or place.  I suppose that I could also elaborate 
on flaws in just about any other fax modem hardware that I've used 
extensively, too.  However, the trick isn't so much finding and using a 
flawless piece of equipment (because there isn't such a thing), but 
rather learning to avoid, fix, and deal with the issues as they appear.  
And this is why I have such a hard time favoring fax Class 2 and to a 
greater extent fax Class "proxies" like the Diva Server... because they 
take away so much of the control that is necessary to avoid, fix, and deal.


My hope in IAXmodem is merely to extend that aspect of control into the 
DCE and DSP.


Lee.

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Re: Re: [Asterisk-Users] E400P vs te410p vs te411p

[Asterisk]

Good Day,

Varion will happily beat pbxhardware.com's price for the Tormenta II for Ebay 
Customers,
Please see http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=5818734724

Sincerely,
Varion-Inc.



- Original Message -
From: MvPhone
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: Re: [Asterisk-Users] E400P vs te410p vs te411p
Sent: 10/12/2005 2:32:02 PM

Hi,

Check out http://store.pbxhardware.com => it has better prices on the
E400P / T400P cards. There are also 2 port versions of these.

>Hi,
>
>I found E400P quad PRI card quite cheap (749USD):
>
>http://www.govarion.com/product_info.php?cPath=1&products_id=2&osCsid=68cdd6e3d08754
>
>in comparison to te410p (approx 1500 USD )
>
>http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TE410P



This message was checked by MailScan for WorkgroupMail.
www.govarion.com 

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[Asterisk-Users] Wanting to Make a PocketPC have a secure Connection to asterisk server

[Kellner, Peter]

Does anyone know of a good solution to create a secure
(encrypted) connection from a pocketpc (IPAQ 6515 in my case) to an asterisk
server?

 

Thanks

 

Peter Kellner

http://PeterKellner.net

 






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[Asterisk-Users] How can I use different languages (Chinese, Cantoneese)?

[Ronald Wiplinger]

I want to give the users the announcements as they subscribed to. The 
announcements should be in English, Chinese, Cantonese, according to 
their phone number. How can I do that? I can hardly make for each number 
a different context!!!



bye

Ronald Wiplinger



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Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?

[Darren Nickerson]

"Lee Howard" <[EMAIL PROTECTED]>

Well, part of my preference for the 2977s involves my strong dislike for 
the way that the Diva Servers and BrookTrouts do things.  It's enough of a 
dislike to get me over the learning curve of how to properly set up the 
2977s for HylaFAX use.


I agree, there's a lot required to get the 2977s to play nicely, and in fact 
to get them to play in Class 1 at all. I still have nightmares about the 
time we learned the Patton was faking "rings" right on HylaFAX's 6000ms 
intra-ring timer, ... thank heavens for HylaFAX's RingTimeout.


I'm curious though, in an earlier message you wrote:

"Sending is also quite  good, however, there are some quirks with the Patton
firmware which need to be resolved for me to be completely delighted."

Are these issues you refer to just cosmetic 'nice to haves' or do they 
actually impact the outbound success rate? Is the 2977's outbound in Class 1 
as solid as the Mainpine or Multitech boards now? Our experience has shown 
that is not the case ...


If your favor involves V.34 then it may be a while before the relevant 
patents expire.


I know, that's a shame. Still, there will be many cases where V.34 won't be 
such a major loss and IAXmodem will make sense. Can't beat the pricetag! ;-)


-Darren

--
Darren Nickerson
Senior Sales & Support Engineer
iFAX Solutions, Inc. www.ifax.com
[EMAIL PROTECTED]
+1.215.438.4638 x8106
+1.215.243.8335 (fax) 


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Re: [Asterisk-Users] Re: Modifying cmd VoicemailMain

[FELIX E SKOWRONEK]

I have also been looking for a way to customize voicemail (I want to add a 
"pause" feature and change the promps).  I have come to the same conclusions 
as to where to do it, but have not yet created a solution.  I have found 
this posting/forum which gives insight into modifying the "app_voicemail.c" 
file but does not directly address our issues, it's good info however:


http://www.voipuser.org/forum_topic_2952.html

Something tells me it will be a combination of modification, agi and 
dialplan.


There is also another person with the similar goals here:

http://sourceforge.net/forum/forum.php?thread_id=1363794&forum_id=420324

If you examine the app_voicemail.c file it refers to these links/people who 
contributed to different language translations and additions:


* 12-16 - 2005 : Support for Greek added by InAccess Networks (work funded 
by HOL, www.hol.gr)

*George Konstantoulakis <[EMAIL PROTECTED]>
* 05-10 - 2005 : Support for Swedish and Norwegian added by Daniel Nylander, 
http://www.danielnylander.se/

*
* 05-11 - 2005 : An option for maximum number of messsages per mailbox added 
by GDS Partners (www.gdspartners.com)

*Stojan Sljivic <[EMAIL PROTECTED]>
*
* 07-11 - 2005 : An issue with voicemail synchronization has been fixed by 
GDS Partners (www.gdspartners.com)

*Stojan Sljivic <[EMAIL PROTECTED]>

And of course there is [EMAIL PROTECTED] (I'm not ready to bother him yet, 
although someone willing to spearhead a Japanese translation project might 
get some attention or at least a reference).


I would love to hear about anyone's success in this area! And I will be sure 
to post my progress in these threads.


F


From: "trixter http://www.0xdecafbad.com"; <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial 
Discussion
To: Asterisk Users Mailing List - Non-Commercial 
Discussion

Subject: Re: [Asterisk-Users] Re: Modifying cmd VoicemailMain
Date: Wed, 12 Oct 2005 18:18:26 -0700

On Thu, 2005-10-13 at 10:08 +0900, Kuniyoshi Murata wrote:
> Andy Kuo writes:
>
> > Hi,
> >  Maybe you can record the sound file "vm-five.gsm" as "five hour" in
> > Japanese, instead of just "five".
> >  AK
>
> I don't think you can do that.
> Because that vm-five.gsm can be used as message number also (e.g. 
"message FIVE")

>

For the other changes I am starting to think that it will require either
modifying the voicemail app or doing voicemail as an agi or dialplan
setup.  All 3 have some drawbacks, but would give you the ability to
tweak everything exactly how you want it.

As either an agi or dialplan setting you could use most of the voicemail
app functionality if that is suitable (I dont know where the prompts are
exactly that the original poster refered to).  It may boil down to
writing a complete voicemail system as an agi or modifying the voicemail
app to get exactly what is wanted.




_
Don’t just search. Find. Check out the new MSN Search! 
http://search.msn.click-url.com/go/onm00200636ave/direct/01/


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Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?

[Lee Howard]

Darren Nickerson wrote:

We prefer the Eicon Diva server and Brooktrout TR1034 boards, which 
are known to work well with HylaFAX since we've had our share of 
headaches with the 2977's.



Well, part of my preference for the 2977s involves my strong dislike for 
the way that the Diva Servers and BrookTrouts do things.  It's enough of 
a dislike to get me over the learning curve of how to properly set up 
the 2977s for HylaFAX use.


Having said that, I'm excited to see Lee and Steve improving IAXmodem 
and the underlying SpanDSP library, and look forward to the day that 
is performs similarly (or better) to the DSP-laden boards we presently 
favor!



If your favor involves V.34 then it may be a while before the relevant 
patents expire.


Lee.

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[Asterisk-Users] New Application: Broadcast

[Begumisa Gerald M]

Hello,

I've released an Asterisk application under the terms of the GNU GPL.  You
may find it here:

http://psg.com/~begg/projects/

A short exerpt from the README:

--
Broadcast is an Asterisk (http://www.asterisk.org) application which you
may use to send a generic message over TCP/IP to any number of computers
running software configured to listen for these types of messages. Being
written in C, Broadcast will be dynamically loaded onto the Asterisk
program on startup, making it a highly reliable and scalable option when
compared with other solutions based on the Asterisk Gateway Interface
(AGI) system...
--

Hope someone finds it useful!

Cheers,
Gerald.

PS:
Sorry for the cross posts!
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Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?

[Darren Nickerson]

"Lee Howard" <[EMAIL PROTECTED]> wrote:

Yes I have a Patton 2977 (driven by HylaFAX), connected via crossover to 
one port on a TE405P (driven by Asterisk) which has another port connected 
to the T1 from the telco.  Asterisk bridges the two for sending and 
receiving.  Receiving is wonderful.  Sending is also quite good, however, 
there are some quirks with the Patton firmware which need to be resolved 
for me to be completely delighted.


Like Lee, we've had much success using Asterisk as a bridge between the PSTN 
and a HylaFAX server with T1/E1 fax boards. We prefer the Eicon Diva server 
and Brooktrout TR1034 boards, which are known to work well with HylaFAX 
since we've had our share of headaches with the 2977's. If you're looking 
for something that works today with HylaFAX, supports all the important fax 
features including V.34-speed (33.6)faxing, I can recommend the Diva Server 
highly - it comes in 8 and 24-port PRI, and 2 and 8-port BRI flavors.


Having said that, I'm excited to see Lee and Steve improving IAXmodem and 
the underlying SpanDSP library, and look forward to the day that is performs 
similarly (or better) to the DSP-laden boards we presently favor!


Tom, if you really only have a single-port PRI and can't shell out for a 
dual, then a T1 fax board is out of the question - it's even more cash. That 
doesn't leave you with a lot of options except to outsource your faxing. Why 
not give Lee's stuff a try some weekend when your system's idle?


-Darren

--
Darren Nickerson
Senior Sales & Support Engineer
iFAX Solutions, Inc. www.ifax.com
[EMAIL PROTECTED]
+1.215.438.4638 x8106
+1.215.243.8335 (fax) 


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Re: [Asterisk-Users] Re: Modifying cmd VoicemailMain

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

On Thu, 2005-10-13 at 10:08 +0900, Kuniyoshi Murata wrote:
> Andy Kuo writes: 
> 
> > Hi,
> >  Maybe you can record the sound file "vm-five.gsm" as "five hour" in
> > Japanese, instead of just "five".
> >  AK
> 
> I don't think you can do that.
> Because that vm-five.gsm can be used as message number also (e.g. "message 
> FIVE") 
> 

For the other changes I am starting to think that it will require either
modifying the voicemail app or doing voicemail as an agi or dialplan
setup.  All 3 have some drawbacks, but would give you the ability to
tweak everything exactly how you want it.  

As either an agi or dialplan setting you could use most of the voicemail
app functionality if that is suitable (I dont know where the prompts are
exactly that the original poster refered to).  It may boil down to
writing a complete voicemail system as an agi or modifying the voicemail
app to get exactly what is wanted.


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


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Re: [Asterisk-Users] Soekris and Asterisk

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

On Wed, 2005-10-12 at 17:46 -0700, Paul Mahler wrote:
> You need about 30MHz per channel. That means the Soekris can only handle part
> of a T1, it will never handle a quad span. 
> 
> Paul
> 

How was that determined?  

I have a problem with a plain number like that, which may have been
taken into account, why I am asking...  

Different cpus operate differently, taking more or less time to complete
certain functions.  Instruction optimization can go a long way if those
instructions are used (not terribly likely if its just pushing bits but
there are some for just that).

Additionally there is no codec processing (presumably) with TDMoE, does
the 30MHz take into account any codec processing or is it literally
30MHz (on what cpu class?!) for just pushing bits?

There are other factors, but you did say 'about' so they are optional to
this conversation, ie other IRQs on the box, potential for device
polling, etc.  A tuned system for that specific task (pushing bits
between a TDM card and ethernet via TDMoE) may be able to operate at a
lower clock speed per channel, but that isnt as important for the
initial questions.



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


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[Asterisk-Users] Re: Modifying cmd VoicemailMain

[Kuniyoshi Murata]

Andy Kuo writes: 


Hi,
 Maybe you can record the sound file "vm-five.gsm" as "five hour" in
Japanese, instead of just "five".
 AK


I don't think you can do that.
Because that vm-five.gsm can be used as message number also (e.g. "message FIVE") 


--
Kuniyoshi Murata.iChat/AIM:macwebcaster
English-Japanese Interpreter mailto:[EMAIL PROTECTED]
Macintosh Webcast Specialisthttp://www.macwebcaster.com 
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Re: [Asterisk-Users] Modifying cmd VoicemailMain

[Andy Kuo]

Hi,
 
Maybe you can record the sound file "vm-five.gsm" as "five hour" in Japanese, instead of just "five".
 
AK 
On 10/12/05, Kuniyoshi Murata <[EMAIL PROTECTED]> wrote:
Dear Asterisk Users,I'm a Japanese and now configuring Voicemail.Now I need to modify the way cmd VoicemailMain works to fix language
difference and other my conveniences.What I want to do are...1) Add words used in message retrieving guidance.I need to add different suffixes to numeric words due to Japanese way ofmentioning time. (
e.g. in English, you can say "Five forty-five" for 5:45,but in Japanese, we have to put "hour" and "minute" for respective time unit(meaning, VoicemailMain should pronounce as "Five hours and forty-five
minutes" in Japanese). So, is there any way to add words modifying theregular word order?2) Disable most of the key function guidance for retrieving the message.I don't want too much function guidance of VoicemailMain saying such as "3
for advanced options" and the like. I just want to hear just a few importantkeys to press. So, is there any way I can separately disable guidance foreach key functionsAny input is welcome.
TIAKuni--Kuniyoshi Murata.iChat/AIM:macwebcasterEnglish-Japanese Interpreter mailto:[EMAIL PROTECTED]Macintosh Webcast Specialist
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Re: [Asterisk-Users] Soekris and Asterisk

[Paul Mahler]

You need about 30MHz per channel. That means the Soekris can only handle part
of a T1, it will never handle a quad span. 

Paul

--- Kristian Kielhofner <[EMAIL PROTECTED]> wrote:

> Craig Guy wrote:
> > Has anyone on the list used a Soekris engineering PC as a TDM - Ethernet 
> > bridge?  For example something like a net4801 with a TE110p in it and 
> > then using TDMoE to get it into a bigger server where the call 
> > processing proper will occur.
> > 
> > Anyone know if it might handle a quadspan card ok? (no transcoding, just 
> > pure PRI to TDMoE bridging).
> > 
> > Craig
> 
> Craig,
> 
>   It all depends on where you are going to do what (PRI, echo cancel, 
> etc).  Also, for four spans the interrupt load alone could probably 
> saturate the CPU.
> 
>   If you want to try, AstLinux will be an excellent start...
> 
> http://www.astlinux.org
> 
> P.S. - I created AstLinux, so of course I would recommend it!
> 
> --
> Kristian Kielhofner
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> 


Paul Mahler
[EMAIL PROTECTED]
www.signate.com
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Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?

[Lee Howard]

trixter http://www.0xdecafbad.com wrote:


Perhaps Lee can comment on exactly how 'development grade' it really is,
perhaps even cite some test cases where people have used it on larger
scale operations (ie larger than a home users 1-2 times a month or
less).



According to sourceforge, there have currently been 68 downloads of the 
software.  As there have been four releases, I suspect that many of 
those were repeat offenders coming back for updates.


IAXmodem is merely a bridge between libiax2 and spandsp (the DSP part, 
not the txfax/rxfax Asterisk applications).  Both libiax2 and spandsp 
have been through years of exposure already, and those are, for the most 
part, to be considered production-grade code.  IAXmodem does use a 
relatively new feature in spandsp (the T.31 DCE) to produce a Class 1 
modem, and that part has had relatively little exposure in comparison 
to, say, the V.29 modem in spandsp.


In *my* usage of IAXmodem I've sent and received hundreds of faxes with 
it in development between IAXmodem and regular modems and IAXmodem and 
fax machines.  In my usage of IAXmodem in "real world" production 
purposes I've sent probably a few hundred faxes and received maybe a 
little less than a hundred.


I've not entirely dumped my Patton 2977 installations for mostly one 
major reason:  spandsp doesn't yet support V.17 (14400 bps) fax 
reception very well.  V.29 seems fine for the most part (I am 
troubleshooting the one exception that I know of).  Once spandsp 
supports V.17 fax reception well I'll then be on a fast track to 
replacing my Patton installations and using IAXmodem instead.  At that 
point I'll be able to say it's "production-grade" code because I'll 
actually be using full-blown on production systems.


We must not confuse "developer-grade" with "unstable".  And 
"production-grade" does not mean "stable", either.  Currently IAXmodem 
is "developer-grade" because it's simply not had enough exposure for it 
to be called otherwise.  Also, the documentation is only to be found 
within the README and other files within the tarball and there probably 
is some "polish" that I'd like to see done before I start promoting it 
as "production-grade".  But, none of this is due to any "instability".  
(I interpret "unstable" to mean that the software is known to have 
unpredictable results.  I would never have even released IAXmodem 
publicly in the first place if this were the case.)


Lee.

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[Asterisk-Users] Notice message meaning for C7960?

[Rich Adamson]

Asterisk cvs-head compiled 2005-10-07 11:

Oct 12 18:35:12 NOTICE[21740]: chan_sip.c:10685 handle_request_register: Registr
ation from 'sip:[EMAIL PROTECTED]' failed for '208.5.218.28' - Not a lo
cal SIP domain

The sip phone is a Cisco 7960 with one line defined, and registration
with * is occuring just fine. Calls to/from the phone are fine. The
phone is on a distant registered IP address. Two years of running *, but not 
seen this one before. Several other C7960's don't generate this warning 
message. Obviously, I've screwed something up. 

What did I do or miss? Cluebat?

Rich


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Re: [Asterisk-Users] send Q931 information element keypadfacility ?!

[Matthew Fredrickson]

On Wed, Oct 12, 2005 at 08:43:32PM +0200, Bruno Voigt wrote:
> I'm looking for a way with any asterisk-version with TE410P (cpe
> EuroISDN, Q931)
> for sending an INFORMATION ELEMENT KeypadFacility,
> eg. *87, during a connected call to the PSTN switch.
> 
> Are there existing functions in asterisk to generate & send such IE ?
> 
> If not what existing modules would be best to derive from?

Not currently.

-- 
Matthew Fredrickson
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Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?

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

On Wed, 2005-10-12 at 19:19 -0400, Tom Rymes wrote:
> > Would that not solve in the short term all of those issues or am I
> > missing something?
> 
> Well, I can redirect a DID to it, but I have no fax traffic going to  
> that DID, and I am still reluctant to install "developer-grade" code  
> on my production asterisk server.
> 
The idea of redirecting an unused did is so that you can develop your
test cases then see if the code works how you expected it.  I would hope
that you wouldnt have any real traffic aside from your test cases :)

As for development code, I can understand that, and is actually a good
practice to only use stable stuff...  However remember that it is open
source and often it stays in development much longer than most companies
selling a product keep code in the dev stages.  This is because its not
being sold so there isnt market pressure to make it 'stable'.  Far too
often commercial products (not all but enough) release 'stable' products
that are far from it, infact they act more like they are in the final
beta stages ...  

Perhaps Lee can comment on exactly how 'development grade' it really is,
perhaps even cite some test cases where people have used it on larger
scale operations (ie larger than a home users 1-2 times a month or
less).


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


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Re: [Asterisk-Users] TDM04B card with only 3 lines connected using chanisavail

[Rich Adamson]

> I noticed that using a TDM04B and only having 3 analog lines connected 
> at this time (4th is coming)
> I was using ChanIsAvail(Zap/4&Zap/3&Zap/2&Zap1) which gives me an 
> available line no problem.
> 
> However Zap/3 did not actually have a line connected at this time. Yet 
> asterisk still gave me it as
> an option for dialout even though no dial tone or anything
> 
> Is that expected... Obviously I can edit zapata and not use it. but I 
> just thought asterisk would
> potentially detect no line connected and not even give me it for 
> chanisavail?

Nope, there is no code to detect an unconnected pstn line. The chipset
has far more capabilities for such stuff then has been implemented,
but obviously those functions have not been addressed. (Note: there
was, and maybe still is, a compile-time option to detect unconnected
pstn lines, but not sure the option still exists.)

Per the chipset spec sheets, the TDM card can detect bridged handsets,
unconnected pstn lines, reverse polarity, and much more. Adding the
code to support that stuff is not a trevial task given the different
country standards that should be addressed, etc.



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Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?

[Tom Rymes]

On Oct 12, 2005, at 6:58 PM, trixter http://www.0xdecafbad.com wrote:


On Wed, 2005-10-12 at 18:45 -0400, Tom Rymes wrote:


This is true, but:

1.) Lee has stated that IAXModem is still "Developer-grade" code.
2.) I don't have a spare PRI for testing, and our phone system is far
too mission critical for me to go mucking about with it and trying
this out (especially given #1, above).
3.) It will not be easy for me to test out this setup without simply
switching our production HylaFAX server to use IAXModem, which I am
again reluctant to do, seeing as it is our production server and we
depend on it. Testing fax service setups is notoriously difficult due
to the huge number of different fax machines, etc that are out there.


redirect one did to iaxmodem for now, test it out, you shouldnt  
have to

reconfigure everything to get this to work.  iaxmodem connects via iax
so it acts like any other client in that regard, so you should just  
set

up an acct and redir an unused did to it, assuming you have an unused
did of course.


Would that not solve in the short term all of those issues or am I
missing something?


Well, I can redirect a DID to it, but I have no fax traffic going to  
that DID, and I am still reluctant to install "developer-grade" code  
on my production asterisk server.


I'm sure it would be OK, and I bet it would work great, but frankly I  
would rather wait for the code to develop further and give some  
others a chance to figure it out at first. I'm not a big fan of  
running my production servers on the bleeding edge, and I don't have  
a spare PRI to use for testing on a non-production basis. (got a  
spare $400 per month or so?)


Tom
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Re: [Asterisk-Users] parameters documentation

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

On Wed, 2005-10-12 at 09:41 -0400, Time Bandit wrote:
> > Is there a "place" where all the parameters are documented ?
> > In example (my example!) I would like to know the meaning of a lot of
> > parameter that can be used in sip.conf,
> 
> http://www.voip-info.org/wiki-Asterisk+config+sip.conf
> 
> How did I found this ?
> 
> http://www.google.ca/search?hl=en&q=site%3Avoip-info.org+sip.conf&btnG=Google+Search&meta=
> 
> Remember : google is your friend


to elaborate slightly ... if you type into google
site:voip-info.org asterisk  
where type is cmd or config
and item is either the config file name or the command you should be
able to get there.  Alternatively you can just straight there by
entering the url:

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


Google is handy if you dont know the name of the command in question
because you can just omit  and it will show all the commands
available :)

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Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?

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

On Wed, 2005-10-12 at 18:45 -0400, Tom Rymes wrote:
> This is true, but:
> 
> 1.) Lee has stated that IAXModem is still "Developer-grade" code.
> 2.) I don't have a spare PRI for testing, and our phone system is far  
> too mission critical for me to go mucking about with it and trying  
> this out (especially given #1, above).
> 3.) It will not be easy for me to test out this setup without simply  
> switching our production HylaFAX server to use IAXModem, which I am  
> again reluctant to do, seeing as it is our production server and we  
> depend on it. Testing fax service setups is notoriously difficult due  
> to the huge number of different fax machines, etc that are out there.
> 
redirect one did to iaxmodem for now, test it out, you shouldnt have to
reconfigure everything to get this to work.  iaxmodem connects via iax
so it acts like any other client in that regard, so you should just set
up an acct and redir an unused did to it, assuming you have an unused
did of course.


Would that not solve in the short term all of those issues or am I
missing something?

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Re: [Asterisk-Users] parameters documentation

[Rich Adamson]

I'll second that... good reading so far, have not finished it though.

Rich



> The people who have been documenting Asterisk have been working on a book 
> for the last few months, it has been published by O'reilly (Asterisk-The 
> Future of Telephony)and is just now finding it's way into the major 
> bookstores, listed under Open-Source at Barns&Noble.
> 
> While it will not answer everything asterisk can do, but its glossery and 
> and appendix are very helpful for quick reference.  If you have been 
> following the Asterisk Documentation Project, some of it will be old hat, 
> but I'm looking forward to replacing my huge stack of printouts with it.
> 
> It gives a pretty good overview of VOIP, Networking, Telephony, etc.
> 
> http://www.oreilly.com/catalog/asterisk/
> 
> 
> >From: "Steve Totaro" <[EMAIL PROTECTED]>
> >Reply-To: Asterisk Users Mailing List - Non-Commercial 
> >Discussion
> >To: "Asterisk Users Mailing List - Non-Commercial 
> >Discussion"
> >Subject: Re: [Asterisk-Users] parameters documentation
> >Date: Wed, 12 Oct 2005 11:17:01 -0400
> >
> >There is plenty of documentation online for both the 3com and *.  You have
> >to have good search skills I guess.
> >
> >3com has the best knowledge base I have seen.
> >http://knowledgebase.3com.com/ and there are tons of 3com dealers that can
> >help.
> >
> >I think you may need to learn some basic networking before learning
> >asterisk.  NAT is a very basic concept in networking as well as ports such
> >as 5060 (standard port for SIP).
> >
> >There is a very steep learning curve for asterisk and networking in 
> >general.
> >If you want to learn it then you need to dig into the wiki and read all the
> >posts that come across the user's list (well maybe not all  of them).
> >
> >There are plenty of consultants that you can hire if you are not up to it.
> >
> >
> >- Original Message -
> >From: <[EMAIL PROTECTED]>
> >To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> >
> >Cc: "Asterisk Users Mailing List - Non-Commercial Discussion"
> >; 
> ><[EMAIL PROTECTED]>
> >Sent: Wednesday, October 12, 2005 10:34 AM
> >Subject: Re: [Asterisk-Users] parameters documentation
> >
> >
> > > I come from a NBX100
> > > No documentation available.
> > > 1 day it starts saying: "syslog full" and voicemail stop working
> > > No one was able to tell me what was the meaning of that alert
> > > .
> > > 3COM NBX anyway is a good product, but the price is too high, especially 
> >4
> > > years ago, and especially the price of the telephone is very high.
> > >
> > > Andrea
> > >
> > >
> > >
> > >
> > >
> > >  "asterisk"
> > >  <[EMAIL PROTECTED]
> > >  echnologies.com>   
> >To
> > >  Sent by:  "Asterisk Users Mailing List -
> > >  asterisk-users-bo Non-Commercial Discussion"
> > >  [EMAIL PROTECTED] 
> > >  m.com  
> >cc
> > >
> > >
> >Subject
> > >  13/10/2005 16.13  Re: [Asterisk-Users] parameters
> > >documentation
> > >
> > >  Please respond to
> > >   Asterisk Users
> > >   Mailing List -
> > >   Non-Commercial
> > > Discussion
> > >  <[EMAIL PROTECTED]
> > >  ists.digium.com>
> > >
> > >
> > >
> > >
> > >
> > >
> > > "> I really hope this project will be implemented, without documentation
> > > evrything is too hard"
> > >
> > > Not for the thousands of people that have figured it out.
> > >
> > > 3Com NBX might be more your speed and plenty of documentation.
> > >
> > >
> > >
> > > > Really strange answer. I am non used to search on playboy.com.
> > > >
> > > > Anyway, if you try to search
> > > > insecure=very
> > > > on www.voip-info.org, you find 742 links , a bit more for me. (I just
> > > want
> > > > to know what it means)
> > > >
> > > > Moreovere, the first 20 links are non accessible at all
> > > >
> > > >
> > >
> >http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+sip+insecure&diff=6
> > >
> > > >
> > > > they speak about tiki-pagehistory.php, which appears not to exist.
> > > >
> > > > no other comments about this.
> > > > 
> > > >
> > > > I know about one project , "asterisk documentation project"
> > > >
> > > > http://www.asteriskdocs.org
> > > >
> > > > in its home page, the first line is
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >  Great software needs great documentation.
> > > >
> > > >
> > > >
> > > >
> > > > I really hope this project will be implemented, without documentation
> > > > evrything is too hard
> > > >
> > > > Andrea
> > > >
> > > >
> > > >
> > > >
> > > >  "Steve Totaro"
> > > >  

Re: [Asterisk-Users] parameters documentation

[Time Bandit]

> Is there a "place" where all the parameters are documented ?
> In example (my example!) I would like to know the meaning of a lot of
> parameter that can be used in sip.conf,

http://www.voip-info.org/wiki-Asterisk+config+sip.conf

How did I found this ?

http://www.google.ca/search?hl=en&q=site%3Avoip-info.org+sip.conf&btnG=Google+Search&meta=

Remember : google is your friend
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Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?

[Tom Rymes]

On Oct 12, 2005, at 6:33 PM, trixter http://www.0xdecafbad.com wrote:


On Wed, 2005-10-12 at 18:18 -0400, Tom Rymes wrote:


On Oct 12, 2005, at 11:26 AM, Lee Howard wrote:


If your PRI comes in to a TE405P or somesuch then you can pass fax
DIDs out through another port on the TE405P and out to a T1
faxmodem (such as a Patton 2977) or a T1 channel bank and then to
analog modems.



Good call, Lee. Unfortunately, we only have a single port Sangoma
card in our asterisk server. In order to do what you suggest, I would
have to buy a dual port card and a channel bank or T1 modem. Thats
more $$$ than is warranted by our fax traffic.

Also, given reports of problems related to frame-slippage and other
weirdness encountered when sending data/fax through Asterisk, I'm
reluctant to invest that money. Have you tried this setup yourself?



Cant iaxmodem work by having asterisk bridge the pri channel as needed
(did based routing perhaps) and then have hylafax use iaxmodem as the
modem it uses.  That should result in no additional hardware, which
means testing can happen with little cost to see if it works for you.

As I understand it iaxmodem just acts like a modem, and doesnt  
actually

do the processing that hylafax does, so the two would work together
instead of one or the other.  I may be wrong on this, but that is the
way it looks to me so far.


This is true, but:

1.) Lee has stated that IAXModem is still "Developer-grade" code.
2.) I don't have a spare PRI for testing, and our phone system is far  
too mission critical for me to go mucking about with it and trying  
this out (especially given #1, above).
3.) It will not be easy for me to test out this setup without simply  
switching our production HylaFAX server to use IAXModem, which I am  
again reluctant to do, seeing as it is our production server and we  
depend on it. Testing fax service setups is notoriously difficult due  
to the huge number of different fax machines, etc that are out there.


Tom

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Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?

[Lee Howard]

trixter http://www.0xdecafbad.com wrote:


Cant iaxmodem work by having asterisk bridge the pri channel as needed
(did based routing perhaps) and then have hylafax use iaxmodem as the
modem it uses.  That should result in no additional hardware, which
means testing can happen with little cost to see if it works for you.

As I understand it iaxmodem just acts like a modem, and doesnt actually
do the processing that hylafax does, so the two would work together
instead of one or the other.  I may be wrong on this, but that is the
way it looks to me so far.



This is all correct.

Lee.
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Re: [Asterisk-Users] ACD/queues question

[Tom Rymes]

On Oct 12, 2005, at 1:30 PM, Pedro Nunes wrote:

Hi there,

Does anyone know how to setup an overflow queue? When a call rings  
on the queue A, if all agents were busy, the call goes to the queue B.


If all agents in queue B were busy, then the call stays on both  
queues until somebody answers it.


I think this is a basic ACD feature available on most PBX that  
support ACD functionality.


Does anybody knows how to do it with asterisk??

Thanks in advance

 Pedro Nunes
What we have done is to set up a single queue that all calls come  
into. For the agents that we want to be our "Front Line" (i.e.:  
Customer Service Reps), we give them a penalty of 0. Our "Overflow"  
group (i.e.: Customer service reps who are also dealing with walk-in  
customers and therefore should not be bothered unless we're really  
busy) gets a penalty of 1, and our "Last Resort" (i.e.: Everyone  
else) people get a penalty of 2.


That way, all of the calls are answered by our front line people,  
unless they are all busy/unavailable. Then, and only then, the calls  
start going to our overflow people, and if they are also all  
unavailable, the calls go to our last resort people. Seeing as how we  
have more than 23 people between the three groups, there should  
technically be no waiting on hold in the queue, even with the PRI  
saturated.


I don't know if this is what you are looking for, but it works  
extremely well for us. To whomever coded this feature, THANK YOU!


To set this up, just edit the queues.conf file and add the penalty to  
each agent's  "member =>" line like this:


; Front-line - Penalty of 0
member => 100,0
; Overflow - Penalty of 1
member => 101,1
;Last Resort - Penalty of 2
member => 102,2

Hope that proves useful to someone

Tom

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Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?

[Lee Howard]

Tom Rymes wrote:


On Oct 12, 2005, at 11:26 AM, Lee Howard wrote:


Tom Rymes wrote:



(I would like to be able to receive faxes reliably
over our PRI)

Until then, however, I still recommend HylaFAX.



If your PRI comes in to a TE405P or somesuch then you can pass fax  
DIDs out through another port on the TE405P and out to a T1  faxmodem 
(such as a Patton 2977) or a T1 channel bank and then to  analog modems.



Good call, Lee. Unfortunately, we only have a single port Sangoma  
card in our asterisk server. In order to do what you suggest, I would  
have to buy a dual port card and a channel bank or T1 modem. Thats  
more $$$ than is warranted by our fax traffic.



IAXmodem then ;-)  As long as you're okay with V.29 (9600 bps).  Keep a 
back-up analog line around attached to a hardware modem until you're 
completely certain that IAXmodem is working perfectly for your needs.


Also, given reports of problems related to frame-slippage and other  
weirdness encountered when sending data/fax through Asterisk, I'm  
reluctant to invest that money. Have you tried this setup yourself? 



Yes I have a Patton 2977 (driven by HylaFAX), connected via crossover to 
one port on a TE405P (driven by Asterisk) which has another port 
connected to the T1 from the telco.  Asterisk bridges the two for 
sending and receiving.  Receiving is wonderful.  Sending is also quite 
good, however, there are some quirks with the Patton firmware which need 
to be resolved for me to be completely delighted.


I've faxed with IAXmodem through that same TE405P without any troubles.  
I've also faxed with IAXmodem through an X100P without troubles.  I've 
never used any of the TDM cards from Digium.  Lots of people seem to 
report other kinds of fax failures with them, and I don't know more than 
that to comment on it.  People have talked about frame slips a lot, and 
I'm not really sure what to make of it.  My understanding is that a 
frame slip causes a momentary audio disruption (as far as the audio 
stream goes).  So I would think that as long as it's not happening at 
inopportune moments or in a frequent manner that a single frame slip 
shouldn't really take down a fax session.  But, I'm sure that it would 
depend on the software.  As far as HylaFAX goes, it wouldn't notice the 
frame slip.  The modem would, and it would probably report it as corrupt 
data or carrier loss... from which both are easily recoverable.


Lee.

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Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?

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

On Wed, 2005-10-12 at 18:18 -0400, Tom Rymes wrote:
> On Oct 12, 2005, at 11:26 AM, Lee Howard wrote:
> > If your PRI comes in to a TE405P or somesuch then you can pass fax  
> > DIDs out through another port on the TE405P and out to a T1  
> > faxmodem (such as a Patton 2977) or a T1 channel bank and then to  
> > analog modems.
> 
> Good call, Lee. Unfortunately, we only have a single port Sangoma  
> card in our asterisk server. In order to do what you suggest, I would  
> have to buy a dual port card and a channel bank or T1 modem. Thats  
> more $$$ than is warranted by our fax traffic.
> 
> Also, given reports of problems related to frame-slippage and other  
> weirdness encountered when sending data/fax through Asterisk, I'm  
> reluctant to invest that money. Have you tried this setup yourself?

Cant iaxmodem work by having asterisk bridge the pri channel as needed
(did based routing perhaps) and then have hylafax use iaxmodem as the
modem it uses.  That should result in no additional hardware, which
means testing can happen with little cost to see if it works for you.

As I understand it iaxmodem just acts like a modem, and doesnt actually
do the processing that hylafax does, so the two would work together
instead of one or the other.  I may be wrong on this, but that is the
way it looks to me so far.


-- 
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UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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[Asterisk-Users] ASTCC and Asterisk 1.2?

[Nate Kapi]

Does everything with AstCC work properly under Asterisk 1.2?
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Re: [Asterisk-Users] Multiple IAX listeners?

[Time Bandit]

> My question is, will this support more than 1 simultaneous connection from
> the same outside IP address, or will only one soft phone function?
>
> or, put another way:
>
> Can multiple soft phones (running on separate computers) be used
> simultaneously from the same outside IP address?

I've successfully tested the following scenario :
- 3 IAX softphones on 3 different computers all on the same LAN
- this LAN was behind a router, which was behind a router, which was
behind a router, which was connected to the internet
- Asterisk was behind a router with port 4569 forwarded to it

so it would look like this :
3 IAX phones --> router --> router --> router --> Internet --> router
-> Asterisk

each phone could call each-other, could call Asterisk, Asterisk could
call each phone

So, the answer is a definite YES

After this test, I was totally sold on IAX and forgot completely about
SIP, except for local LAN

hth
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Re: [Asterisk-Users] Asterisk logo

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

On Wed, 2005-10-12 at 08:59 +, Andrew Nowrot wrote:
> Hi,
> 
> I was wondering if I could use Asterisk logo in my PBX system which I
> want to introduce in my local market. Does anyone know if I must fill
> some legal issues which let me use this logo.
> 
> Best regards

digium is the owner of that, they are revamping (may have completed
that) the document which describes when and where you can use it.

You should contact digium directly to see if they are ready with their
new terms for use (the logo is trademarked).


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Re: [Asterisk-Users] Multiple IAX listeners?

[Andy Hamilton]

> Can multiple soft phones (running on separate computers) be used
> simultaneously from the same outside IP address?

Yep, should work fine.
Consider how any one webserver can handle multiple http requests to port 80.

Or consider when what happens when multiple users behind the same nat
firewall (it will look to the outside world that they will all be the
same public IP) access the same web resources simultaneously...it
works fine.

-a
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Re: [Asterisk-Users] Asterisk logo

[Tom Rymes]

Digium has stated that you need their permission to use the logos.  
However, I was under the impression that if you attributed the  
copyright to them that you would not need their permission. Of  
course, I am not a copyright lawyer ... (IANACL?)


In other words: "Asterisk and the Asterisk logo are copyrights of  
Digium, Inc. " This is what you often see in press releases, ads,  
etc, so why is Asterisk different?


Is there a lawyer in the house?

Tom

On Oct 12, 2005, at 4:59 AM, Andrew Nowrot wrote:


Hi,

I was wondering if I could use Asterisk logo in my PBX system which I
want to introduce in my local market. Does anyone know if I must fill
some legal issues which let me use this logo.

Best regards

Andrew
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Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?

[Tom Rymes]

On Oct 12, 2005, at 11:26 AM, Lee Howard wrote:


Tom Rymes wrote:



(I would like to be able to receive faxes reliably
over our PRI)

Until then, however, I still recommend HylaFAX.


If your PRI comes in to a TE405P or somesuch then you can pass fax  
DIDs out through another port on the TE405P and out to a T1  
faxmodem (such as a Patton 2977) or a T1 channel bank and then to  
analog modems.


Good call, Lee. Unfortunately, we only have a single port Sangoma  
card in our asterisk server. In order to do what you suggest, I would  
have to buy a dual port card and a channel bank or T1 modem. Thats  
more $$$ than is warranted by our fax traffic.


Also, given reports of problems related to frame-slippage and other  
weirdness encountered when sending data/fax through Asterisk, I'm  
reluctant to invest that money. Have you tried this setup yourself?


Tom
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Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?

[Tom Rymes]

On Oct 12, 2005, at 3:44 PM, Bob Goddard wrote:


Asterisk's faxing capabilities are not nearly as advanced, stable, or
easy to set up as HylaFAX. Also, there seem to be many problems with
frame slipping and the like that screw up faxing over Digium  
cards, and

maybe others as well.



Does Hylafax do software based faxing? As far as I knew, it has always
required a DSP.


Lee would be the guy to ask, seeing as he is involved (deeply  
involved, I think) in HylaFAX development. AFAIK, HylaFAX does  
require a DSP, but IAXModem is designed to function as the DSP,  
thereby allowing HylaFAX to take calls that come in via asterisk  
without additional hardware (such as a T1 fax-modem, a crossover  
cable and a multi-port T1 card, as Lee suggested.).



Either way, I was just saying that grabbing a good modem (see HylaFAX
list archives for suggestions - NOT USRobotics!!!) and installing
HylaFAX would be easier, more reliable, and all-in-all, a better
solution than messing with Asterisk's built-in fax capability.


In other words, don't use a soft fax.


No, software vs hardware isn't the point. As Lee mentioned, there are  
many problems with the hardware side of things, too. What I am  
asserting is that Asterisk's fax capabilities are an inferior choice  
to HylaFAX at this time. Maybe in a year or two that will no longer  
be the case. Also, once Lee has some time to develop IAXModem  
further, I bet it will be a great option, and it is software based.


Tom

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Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?

[Lee Howard]

trixter http://www.0xdecafbad.com wrote:


There is very little info on the sf.net page regarding its
capabilities ...  
 



Right now that's intentional.  I still consider it "developer-grade" 
code.  That said, I do use it on small production usage, and it's fine 
there with a few known issues that I'm working on that probably won't be 
of consequence to most.


Does it only do fax or does it do other data communications?  
 



Only fax.

What fax protocols are supported?  
 



Class 1 and Class 1.0 - V.27ter, V.29, and partial V.17 (all as provided 
by spandsp - Steve's working on the V.code and I'm working on the T.code).



Does the destination path from asterisk->whatever need to be iax or will
asterisk properly translate to a different medium (eg presumably
iaxmodem does iax to asterisk, then from asterisk does it matter if you
use sip, zap, h.323, whatever ?)  I cant see where it would matter once
it hits asterisk, but stranger things have happened ...



IAXmodem doesn't care what the channel type is beyond Asterisk.  
Realize, though, that for most usages I recommend running IAXmodem on 
the same system as Asterisk, and that's to avoid any audio corruption 
that would normally occur (even in minute amounts) when passing over a 
UDP/IP network.  So unless you're running your SIP or H.323 fax endpoint 
also on the same system or have somehow managed to prevent any audio 
corruption (jitters, etc... you should know that caveat) along that 
path, then the answer to the question is Zap is what you *should* use.  
You can try other things, but you'll most likely just be wasting your time.


Lee.

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[Asterisk-Users] Asterisk logo

[Andrew Nowrot]

Hi,

I was wondering if I could use Asterisk logo in my PBX system which I
want to introduce in my local market. Does anyone know if I must fill
some legal issues which let me use this logo.

Best regards

Andrew
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[Asterisk-Users] TDM04B card with only 3 lines connected using chanisavail

[Jerry Geis]

I noticed that using a TDM04B and only having 3 analog lines connected 
at this time (4th is coming)
I was using ChanIsAvail(Zap/4&Zap/3&Zap/2&Zap1) which gives me an 
available line no problem.


However Zap/3 did not actually have a line connected at this time. Yet 
asterisk still gave me it as

an option for dialout even though no dial tone or anything

Is that expected... Obviously I can edit zapata and not use it. but I 
just thought asterisk would
potentially detect no line connected and not even give me it for 
chanisavail?


Jerry

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Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?

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

On Wed, 2005-10-12 at 16:48 -0500, Tim Litwiller wrote:
> See IAXModem above for the soft DSP.

There is very little info on the sf.net page regarding its
capabilities ...  

Does it only do fax or does it do other data communications?  

What fax protocols are supported?  

Does the destination path from asterisk->whatever need to be iax or will
asterisk properly translate to a different medium (eg presumably
iaxmodem does iax to asterisk, then from asterisk does it matter if you
use sip, zap, h.323, whatever ?)  I cant see where it would matter once
it hits asterisk, but stranger things have happened ...



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


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Re: [Asterisk-Users] migrated to new ver on voip connection vs1 server voicemail problems

[Tom Vile]

Either Permissions on the directory are incorrect or you have no unavail.wav file.On 10/11/05, Andy Goss <
[EMAIL PROTECTED]> wrote:I migrated to a new version of the voip connection vs1 server software
and I am now getting these errors when I try to call my voicemail.  Anythoughts?  The files are there, so I don't get it.Oct 11 19:57:26 WARNING[6587]: format_wav.c:140 check_header: Not a wavfile 49
Oct 11 19:57:26 WARNING[6587]: file.c:418 ast_filehelper: Unable to openfd on /var/spool/asterisk/voicemail/default/5933/unavail.wavOct 11 19:57:26 WARNING[6587]: file.c:804 ast_streamfile: Unable to open/var/spool/asterisk/voicemail/default/5933/unavail (format ulaw): No
such file or di--H. Andy GossNetwork EngineerNetwork Advocates Inc.Main: 502.412.1050DID: 502.992.5933Mobile: 502.387.8216[EMAIL PROTECTED]
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http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony
www.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856
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Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?

[Tim Litwiller]

Bob Goddard wrote:

On Wednesday 12 Oct 2005 14:53, Tom Rymes wrote:

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Bob Goddard
Sent: Tuesday, October 11, 2005 6:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Which asterisk-friendly cards
are fax-capable?

On Tuesday 11 Oct 2005 22:41, Lee Howard wrote:

Tom Rymes wrote:

Frankly, I would recommend that you forget about trying

to fax with


Asterisk. Buy a good Multitech analog modem and install HylaFAX.

Use the right tool for the job!!!

Actually, you can use HylaFAX and Asterisk together.

  https://sourceforge.net/projects/iaxmodem/

Just be certain that your audio path doesn't run over any

lossy medium


(so run IAXmodem on your Asterisk box).

I'll expand on what Tom meant

Use a hardware based DSP for faxing not software based.

Actually Bob, that isn't what I meant. Lee simply suggested a different
way (IAXModem instead of analog modem) of implementing what I meant. I
would still recommend using analog if you can but, if you cannot, use
IAXModem from Lee.


Okay!


Asterisk's faxing capabilities are not nearly as advanced, stable, or
easy to set up as HylaFAX. Also, there seem to be many problems with
frame slipping and the like that screw up faxing over Digium cards, and
maybe others as well.


Does Hylafax do software based faxing? As far as I knew, it has always
required a DSP.


See IAXModem above for the soft DSP.




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[Asterisk-Users] Maximum retries exceeded on call.

[Peter Ankerstål]

I have set up a asterisk-server behind NAT and peers to another asterisk
and uses that one for outgoing calls. I have som clients on my asterisk
and they could register to it well over internet. Not a problem. But when
they try to call me the asterisk-server tells me this:

Oct 12 23:21:38 WARNING[23360]: chan_sip.c:695 retrans_pkt: Maximum retries 
exceeded on call [EMAIL PROTECTED] for seqno 32458501 (Non-critical Response)

Configs can be found at http://www.pulia.nu/~peter/asterisk/

When they call me they can hear me but I get no sound. Weird.
Any Ideas?



-- 
MVH
Peter Ankerstål.
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Re: [Asterisk-Users] MWI integration between Asterisk and Meridian

[kritikus Araklidas]

TKS buddy if i find o develop myself something regarding that i told you.

Cristian.





From: Gary Reuter <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial 
Discussion
To: Asterisk Users Mailing List - Non-Commercial 
Discussion

Subject: Re: [Asterisk-Users] MWI integration between Asterisk and Meridian
Date: Wed, 12 Oct 2005 17:12:29 -0400

I've been wanting to do exactly the same thing, but I believe it's beyond 
my

coding skills.
I think we need a function similar to the SirrixMWI. Some initial code for
MWI exists in libpri, but nothing in the rest of asterisk calls those
functions yet.


On 10/12/05, kritikus Araklidas <[EMAIL PROTECTED]> wrote:
>
> I try to integrate my old PBX Meridian and Asterisk througth a PRI T1 
(I'm

> gonna use only the asterisk voicemail system) but i don't know how to
> integrate the MWI protocol between Asterisk Voicemail and my Nortel
> meridian.
>
> Anyone know what i have to do for that.?
>
> Any idea is appreciated.
>
> Regards.
>
> Cristian.
>
> _
> Don't just search. Find. Check out the new MSN Search!
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Re: [Asterisk-Users] MWI integration between Asterisk and Meridian

[Gary Reuter]

I've been wanting to do exactly the same thing, but I believe it's beyond my coding skills.
I think we need a function similar to the SirrixMWI.  Some initial
code for MWI exists in libpri, but nothing in the rest of asterisk
calls those functions yet.
On 10/12/05, kritikus Araklidas <[EMAIL PROTECTED]> wrote:
I try to integrate my old PBX Meridian and Asterisk througth a PRI T1 (I'mgonna use only the asterisk voicemail system) but i don't know how tointegrate the MWI protocol between Asterisk Voicemail and my Nortel
meridian.Anyone know what i have to do for that.?Any idea is appreciated.Regards.Cristian._Don't just search. Find. Check out the new MSN Search!
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[Asterisk-Users] MWI integration between Asterisk and Meridian

[kritikus Araklidas]

I try to integrate my old PBX Meridian and Asterisk througth a PRI T1 (I'm 
gonna use only the asterisk voicemail system) but i don't know how to 
integrate the MWI protocol between Asterisk Voicemail and my Nortel 
meridian.


Anyone know what i have to do for that.?

Any idea is appreciated.

Regards.

Cristian.

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RE: [Asterisk-Users] Patton SmartNode

[Guido Hecken]

We use the SmartNodes SN1400 and SN2300 as ISDN Gateways in our customer
Asterisk installations and are really happy with them. They run very stable
and you can configure nearly everything. Support from INALP is also great.
With the interface cards for the SmartNode 2300 you should be able to
connect nearly everything to VOIP.

Regards

Guido Hecken
 
> Does anybody have any experience using a Patton SmartNode as a SIP/Telco
> gateway with Asterisk?  They seem really inexpensive and appear to
> support all of the necessary features, but I don't have any experience
> with their products, so I don't know if they are any good.  We are
> currently using a Cisco 2600 w/ PRI card and it works fine, but I was
> looking for someone else as a possible alternative.  Thanks.
> 
> Peder
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Re: [Asterisk-Users] Monitor DTMF problems

[Michael Stearne]

On 10/12/05, Mir <[EMAIL PROTECTED]> wrote:
> Hello
>
> We have discovered a problem with DTMF on Asterisk.
> We have a setup with a T1 from PSTN going into an Asterisk box, and
> then out again on T1 and into a normal PBX (EADS)
>
> We use it to record all calls going to/from the PBX.
>
> The problem is that when we record the calls (with MONITOR command),
> DTMF tones gets obscured, and is not understood in the other end, if
> we dont Monitor, there are no problems.
>
> It sounds like the tones are cut into two, h hard to explain ...
>
> Does this ring a bell at anyone ?
>

It does to me.  We are having problems with receiving DTMF tones and
are under the impression it's our provider.

Michael
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Re: [Asterisk-Users] VoIP Buster stopped working?

[Justin Richards]

Thanks for the confirmation!
 
 
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[Asterisk-Users] Monitor DTMF problems

[Mir]

Hello

We have discovered a problem with DTMF on Asterisk.
We have a setup with a T1 from PSTN going into an Asterisk box, and
then out again on T1 and into a normal PBX (EADS)

We use it to record all calls going to/from the PBX.

The problem is that when we record the calls (with MONITOR command),
DTMF tones gets obscured, and is not understood in the other end, if
we dont Monitor, there are no problems.

It sounds like the tones are cut into two, h hard to explain ...

Does this ring a bell at anyone ?

Michael
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Re: [Asterisk-Users] which hardware should i use??????????

[Paul]

ishtiaq ahmed wrote:

hy all 


i need a suggesion on what hardware should i use for
the following case study

i have five offices each will be having 35 to 45
extensions. if i will be using voip fones for those
extensions( either it is iax or sip ) which one will
be better and cheaper what should i use. all the five
offices will be connected through asterisk servers (
one in each of the offices ). now the confusion is
that is there any hardware needed to connect the voip
fones to the asterisk server( how we can connect them
to asterisk server ). and for outgoing calls to the
pstn network which card should be used. 


plzz guide me thouroly about the hardware. i have
asked a lot of people every one is giving his own
suggestion. so i thought to ask from the official
mailing list. 


i hope that i will be getting a good response.

i live in pakistan.
 


Your voip phones use ethernet. You just need network cabling and switches.

pstn connection depends on what you have for pstn lines.

Maybe somebody in pakistan can provide the consulting you need. Post on 
the -biz list for that. Spend a little money on a local asterisk guru 
before you spend a lot on hardware. I think 5 servers and about 200 
phones is a large deployment to be undertaking without the help of a 
good consultant.




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RE: [Asterisk-Users] which hardware should i use??????????

[Wiley Siler]

I recommend checking the following site...
www.voip-info.org

Lots of info for you there...

By VoIP phones, I think you are meaning "soft phones" which are software
based.
You will need a headset for the PC that runs the software phone.
Usually Logitech or Plantronics at about $50 a headset.

If you want a hardware phone on a budget then Snom and Grandstream are
popular.

The Asterisk servers only need a card if they will be connected to the
PSTN.  So for a bunch of POTS line you want a TDM card or a channel bank
and T1 card.  The T1 card for a real newbie would be the Digium brand.
A more advanced user might consider a Sangoma.

Good luck and hope you are doing well in Pakistan.

Cheers,
Wiley

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ishtiaq
ahmed
Sent: Wednesday, October 12, 2005 1:08 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] which hardware should i use??

hy all 

i need a suggesion on what hardware should i use for the following case
study

i have five offices each will be having 35 to 45 extensions. if i will
be using voip fones for those extensions( either it is iax or sip )
which one will be better and cheaper what should i use. all the five
offices will be connected through asterisk servers ( one in each of the
offices ). now the confusion is that is there any hardware needed to
connect the voip fones to the asterisk server( how we can connect them
to asterisk server ). and for outgoing calls to the pstn network which
card should be used. 

plzz guide me thouroly about the hardware. i have asked a lot of people
every one is giving his own suggestion. so i thought to ask from the
official mailing list. 

i hope that i will be getting a good response.

i live in pakistan.



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[Asterisk-Users] which hardware should i use??????????

[ishtiaq ahmed]

hy all 

i need a suggesion on what hardware should i use for
the following case study

i have five offices each will be having 35 to 45
extensions. if i will be using voip fones for those
extensions( either it is iax or sip ) which one will
be better and cheaper what should i use. all the five
offices will be connected through asterisk servers (
one in each of the offices ). now the confusion is
that is there any hardware needed to connect the voip
fones to the asterisk server( how we can connect them
to asterisk server ). and for outgoing calls to the
pstn network which card should be used. 

plzz guide me thouroly about the hardware. i have
asked a lot of people every one is giving his own
suggestion. so i thought to ask from the official
mailing list. 

i hope that i will be getting a good response.

i live in pakistan.



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Re: [Asterisk-Users] supermicro with asterisk and tdm cards

[Kevin Bockman]

Cory Andrews wrote:
Yeah I should have picked up on that, single PCI Riser in this one, so 1 
card.  I don't know of any 1U solution out there that would give you 3 
PCI slots to work with, I think you'll have to go to a 2U at least to 
achieve this.
I saw the Dell PowerEdge 1850 has 2 PCI-X on separate busses.  That's 
the only one I've ever seen.



Kevin
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Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?

[Bob Goddard]

On Wednesday 12 Oct 2005 14:54, Tom Rymes wrote:
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > trixter http://www.0xdecafbad.com
> > Sent: Tuesday, October 11, 2005 5:09 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Which asterisk-friendly cards
> > are fax-capable?
> >
> > On Tue, 2005-10-11 at 17:01 -0400, Tom Rymes wrote:
> > > Frankly, I would recommend that you forget about trying to fax with
> > > Asterisk. Buy a good Multitech analog modem and install HylaFAX.
> > >
> > > Use the right tool for the job!!!
> >
> > Asterisk may be able to fax better in the somewhat near
> > future.  One of the things holding up T.38 support is the
> > inability for asterisk to switch codecs on the fly.  I am not
> > saying that is the only thing, just one of the things.  Well
> > 1.2 is supposed to have better support in that regard, which
> > means that work on T.38 can happen in a better way in the future.
>
> This is good news. (I would like to be able to receive faxes reliably
> over our PRI)

It will never happen unless the card has DSPs.
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Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?

[Bob Goddard]

On Wednesday 12 Oct 2005 14:53, Tom Rymes wrote:
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > Bob Goddard
> > Sent: Tuesday, October 11, 2005 6:35 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Which asterisk-friendly cards
> > are fax-capable?
> >
> > On Tuesday 11 Oct 2005 22:41, Lee Howard wrote:
> > > Tom Rymes wrote:
> > > > Frankly, I would recommend that you forget about trying
> >
> > to fax with
> >
> > > > Asterisk. Buy a good Multitech analog modem and install HylaFAX.
> > > >
> > > > Use the right tool for the job!!!
> > >
> > > Actually, you can use HylaFAX and Asterisk together.
> > >
> > >   https://sourceforge.net/projects/iaxmodem/
> > >
> > > Just be certain that your audio path doesn't run over any
> >
> > lossy medium
> >
> > > (so run IAXmodem on your Asterisk box).
> >
> > I'll expand on what Tom meant
> >
> > Use a hardware based DSP for faxing not software based.
>
> Actually Bob, that isn't what I meant. Lee simply suggested a different
> way (IAXModem instead of analog modem) of implementing what I meant. I
> would still recommend using analog if you can but, if you cannot, use
> IAXModem from Lee.

Okay!

> Asterisk's faxing capabilities are not nearly as advanced, stable, or
> easy to set up as HylaFAX. Also, there seem to be many problems with
> frame slipping and the like that screw up faxing over Digium cards, and
> maybe others as well.

Does Hylafax do software based faxing? As far as I knew, it has always
required a DSP.

> Either way, I was just saying that grabbing a good modem (see HylaFAX
> list archives for suggestions - NOT USRobotics!!!) and installing
> HylaFAX would be easier, more reliable, and all-in-all, a better
> solution than messing with Asterisk's built-in fax capability.

In other words, don't use a soft fax.
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Re: [Asterisk-Users] parameters documentation

[FELIX E SKOWRONEK]

The people who have been documenting Asterisk have been working on a book 
for the last few months, it has been published by O'reilly (Asterisk-The 
Future of Telephony)and is just now finding it's way into the major 
bookstores, listed under Open-Source at Barns&Noble.


While it will not answer everything asterisk can do, but its glossery and 
and appendix are very helpful for quick reference.  If you have been 
following the Asterisk Documentation Project, some of it will be old hat, 
but I'm looking forward to replacing my huge stack of printouts with it.


It gives a pretty good overview of VOIP, Networking, Telephony, etc.

http://www.oreilly.com/catalog/asterisk/



From: "Steve Totaro" <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial 
Discussion
To: "Asterisk Users Mailing List - Non-Commercial 
Discussion"

Subject: Re: [Asterisk-Users] parameters documentation
Date: Wed, 12 Oct 2005 11:17:01 -0400

There is plenty of documentation online for both the 3com and *.  You have
to have good search skills I guess.

3com has the best knowledge base I have seen.
http://knowledgebase.3com.com/ and there are tons of 3com dealers that can
help.

I think you may need to learn some basic networking before learning
asterisk.  NAT is a very basic concept in networking as well as ports such
as 5060 (standard port for SIP).

There is a very steep learning curve for asterisk and networking in 
general.

If you want to learn it then you need to dig into the wiki and read all the
posts that come across the user's list (well maybe not all  of them).

There are plenty of consultants that you can hire if you are not up to it.


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

Cc: "Asterisk Users Mailing List - Non-Commercial Discussion"
; 
<[EMAIL PROTECTED]>

Sent: Wednesday, October 12, 2005 10:34 AM
Subject: Re: [Asterisk-Users] parameters documentation


> I come from a NBX100
> No documentation available.
> 1 day it starts saying: "syslog full" and voicemail stop working
> No one was able to tell me what was the meaning of that alert
> .
> 3COM NBX anyway is a good product, but the price is too high, especially 
4

> years ago, and especially the price of the telephone is very high.
>
> Andrea
>
>
>
>
>
>  "asterisk"
>  <[EMAIL PROTECTED]
>  echnologies.com>   
To

>  Sent by:  "Asterisk Users Mailing List -
>  asterisk-users-bo Non-Commercial Discussion"
>  [EMAIL PROTECTED] 
>  m.com  
cc

>
>
Subject

>  13/10/2005 16.13  Re: [Asterisk-Users] parameters
>documentation
>
>  Please respond to
>   Asterisk Users
>   Mailing List -
>   Non-Commercial
> Discussion
>  <[EMAIL PROTECTED]
>  ists.digium.com>
>
>
>
>
>
>
> "> I really hope this project will be implemented, without documentation
> evrything is too hard"
>
> Not for the thousands of people that have figured it out.
>
> 3Com NBX might be more your speed and plenty of documentation.
>
>
>
> > Really strange answer. I am non used to search on playboy.com.
> >
> > Anyway, if you try to search
> > insecure=very
> > on www.voip-info.org, you find 742 links , a bit more for me. (I just
> want
> > to know what it means)
> >
> > Moreovere, the first 20 links are non accessible at all
> >
> >
>
http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+sip+insecure&diff=6
>
> >
> > they speak about tiki-pagehistory.php, which appears not to exist.
> >
> > no other comments about this.
> > 
> >
> > I know about one project , "asterisk documentation project"
> >
> > http://www.asteriskdocs.org
> >
> > in its home page, the first line is
> >
> >
> >
> >
> >
> >  Great software needs great documentation.
> >
> >
> >
> >
> > I really hope this project will be implemented, without documentation
> > evrything is too hard
> >
> > Andrea
> >
> >
> >
> >
> >  "Steve Totaro"
> >  <[EMAIL PROTECTED]
> >  echnologies.com>
> To
> >  Sent by:  "Asterisk Users Mailing List -
> >  asterisk-users-bo Non-Commercial Discussion"
> >  [EMAIL PROTECTED] 


> >  m.com
> cc
> >
> >
> Subject
> >  12/10/2005 14.53  Re: [Asterisk-Users] parameters
> >documentation
> >
> >  Please respond to
> >   Asterisk Users
> >   Mailing List -
> >   Non-Commercial
> > Discussion
> >  <[EM

[Asterisk-Users] RE: faxing to/from asterisk - new

[Technical Support]

The fax2mail and 
mail2fax scripts can be found on www.generationd.com
 
 

Michelle 
DupuisTechnical Support SpecialistOxford Consulting Group Ltd.Making IT work for your 
business...
 
T: (519) 672-8238E: 
[EMAIL PROTECTED]W: 
www.ocg.ca 

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[Asterisk-Users] Canadian Association of VoIP Providers

[John Lange]

My apologies for the cross-posting.

If you are a business or individual providing Voice over IP services in
Canada then we encourage you to read this email carefully otherwise
please disregard.

-

As you are most likely aware, the CRTC has undertaken the roll of
regulating VoIP services in Canada and is currently conducting hearings
with the goal of putting in place regulatory requirements for all VoIP
providers.

Specifically, the CRTC's CISC VoIP 911 working group
( http://www.crtc.gc.ca/cisc/eng/cisf3e4_20.htm ) is very actively
looking at what regulations to put in place in order to implement E911
services for VoIP.

The recommendations of this committee will have direct impact on your
business. Currently this working group is is largely comprised by the
Local Exchange Carriers (ILECs & CLECs) with representation from the
large VoIP providers (Primus & Vonage). To date only a very few smaller
VoIP providers are participating.

Subsequently, much of the discussion is oriented around solutions
designed to work in the traditional telco world. Depending on your
companies infrastructure these solutions may be very expensive or
completely impossible for your business to implement.

Some members of the working group are even of the position that VoIP
service be abolished altogether.

Your companies direct participation in the hearings is the best way to
have an impact. However, we acknowledge that not all companies have the
time and/or resources to fully participate lengthy public hearings.

It is with this in mind we propose the formation of a Canadian industry
association for VoIP providers and we invite you to participate.

The short term goal is to contact and organize Canadian VoIP providers
into a formal association.

Longer term the association will work towards the following goals:

- Keep VoIP providers informed about current regulatory issues
- Ensuring VoIP providers have a place at the CRTC table
- Develop industry recommendations
- Communicate industry recommendations to the CRTC working group
- Communicate industry positions to the media
- Other (to be determined by the association)

At the outset it is envisioned that this group would work in the
following way:

- No membership fee
- Regular updates via email list
- Frequent Conference calls
- No face-to-face meetings (no travel)
- Development of an Industry web site
- In-person representation at each CRTC meeting (The CRTC working group
meets monthly in a different province each month. We hope to have at
least one member representative attend each meeting.)

To voice your support (or opposition) for the formation of this group
please contact me directly either by email or telephone (contact
information in the signature).

It is important that you do not delay. CISC working group
recommendations to the CRTC are forthcoming.

You will be contacted with details on how to participate in the
formation of this association. Our intention is to hold our first
conference call as early as possible (early next week).

NOTE: No web site or association material yet exists because the group
has not been officially formed and named. This will be one of the first
items of business for the new group.

Regards,
-- 
John Lange
President OpenIT ltd. www.Open-IT.ca (204) 885 0872
VoIP, Web services, Linux Consulting, Server Co-Location

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[Asterisk-Users] Is it possible to listen and respond on more than one IAX port?

[Steve Gladden]

Hello,

I'd like to know if it is possible to get * to listen and respond on more
than just one single udp port.

I've run into several situations where I'd like IAX to work on an alternate
port as well as be able to work on the standard port.

I'm wondering if there is a way to do this?

Thanks!!

Steve


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RE: [Asterisk-Users] Polycom: Button Remapping, HELP!

[Andy Goss]

It is on page 22 and 23 in my admin guide.

Andy

--
H. Andy Goss
Network Engineer
Network Advocates Inc.
Main: 502.412.1050
DID: 502.992.5933
Mobile: 502.387.8216
[EMAIL PROTECTED]
 

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Matthew T. O'Connor
> Sent: Wednesday, October 12, 2005 2:38 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Polycom: Button Remapping, HELP!
> 
> While I don't have it working yet, I think I have it figured out.  I
> have to add  entries to my sip.conf  Based on your example I
was
> able to find the relevant info in the Polycom SIP 1.5 Admin Guide
> section 4.6.1.15.
> 
> My next question, which I haven't found in the admin guide (at least
not
> yet) is where to you get a list of the buttons and their respective
> numbers?
> 
> Thanks again,
> 
> Matthew
> 
> 
> Mojo with Horan & Company, LLC wrote:
> > Do you already have an  block in your sip.cfg?  add
the
> >  in there:
> > Try putting:
> > 
> >   ...
> >   ...
> >> key.IP_500.31.subPoint.prim="3"/>
> > 
> >
> > Moj
> >
> > Matthew T. O'Connor wrote:
> >> Ok, that would be helpful for me with some other problems, however
I
> >> don't see " >> I'm using the 1.5.2 Sip firmware the the conf files that came with
> >> that, so I don't have an ipmid.cfg file.  Is this something I can
> >> just add to my sip.conf?
> >>
> >> Anyone out there any suggestions on how to do the speed dial
"in-call"?
> >>
> >> Thanks,
> >>
> >> Matt
> >>
> >>
> >>
> >> Mojo with Horan & Company, LLC wrote:
> >>
> >>> Matthew, when I tried this, I couldn't get the soundpoints to dial
> >>> in-call.  They thought there were picking up a new line for a new
> call.
> >>>
> >>> I created a speed-dial entry (in MACADDRESS-directory.xml,
> >>> "Park#70#3") and then in
> >>> ipmid.cfg:
> >>>  >>> key.IP_500.31.subPoint.prim="3"/>
> >>>
> >>> This tells the phone to run Speed Dial 3 whenever the Services
> >>> button (button #31 on a 500/501) is pressed.  I hope someone can
> >>> help us configure them now to dial these digits in-call...
> >>>
> >>> Mojo
> >>>
> >>> Matthew T. O'Connor wrote:
> >>>
>  I need to find a way to have the Polycom phones automatically
park
>  calls.  Right now my users hit #70# (I know the last # is
optional
>  but it speeds it up.) to park a call.  Personally I think this is
>  easy, but my users would like one button to do this for them.
The
>  Polycom has buttons we don't need (Transfer & Services), it would
>  be nice if I could remap one of those buttons to dial #70#.  Or
if
>  I could add a soft button during a call that would work too.
> 
>  Anyone have any suggestions on how to do this?
> 
>  Thanks much,
> 
>  Matthew O'Connor
> 
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> >>>
> >>
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[Asterisk-Users] PPP over ISDN PRI usinf Asterisk

[gshaw]

Hi
Is anyone using Asterisk for PPP over PRI ISDN. Any example would be
appreciated. I saw ZAPRAS and PPPD commands. The documentation has
zaptel.conf example for PPP using T1/E1 clear channel.

Regards
Goutam



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Re: [Asterisk-Users] Polycom: Button Remapping, HELP!

[Mojo with Horan & Company, LLC]

They are in the 1.5 admin guide, pages 22-25

Matthew T. O'Connor wrote:
While I don't have it working yet, I think I have it figured out.  I 
have to add  entries to my sip.conf  Based on your example I was 
able to find the relevant info in the Polycom SIP 1.5 Admin Guide 
section 4.6.1.15.


My next question, which I haven't found in the admin guide (at least not 
yet) is where to you get a list of the buttons and their respective numbers?


Thanks again,

Matthew


Mojo with Horan & Company, LLC wrote:

Do you already have an  block in your sip.cfg?  add the 
 in there:

Try putting:

 ...
 ...
 key.IP_500.31.subPoint.prim="3"/>



Moj

Matthew T. O'Connor wrote:

Ok, that would be helpful for me with some other problems, however I 
don't see "I'm using the 1.5.2 Sip firmware the the conf files that came with 
that, so I don't have an ipmid.cfg file.  Is this something I can 
just add to my sip.conf?


Anyone out there any suggestions on how to do the speed dial "in-call"?

Thanks,

Matt



Mojo with Horan & Company, LLC wrote:


Matthew, when I tried this, I couldn't get the soundpoints to dial 
in-call.  They thought there were picking up a new line for a new call.


I created a speed-dial entry (in MACADDRESS-directory.xml, 
"Park#70#3") and then in 
ipmid.cfg:
key.IP_500.31.subPoint.prim="3"/>


This tells the phone to run Speed Dial 3 whenever the Services 
button (button #31 on a 500/501) is pressed.  I hope someone can 
help us configure them now to dial these digits in-call...


Mojo

Matthew T. O'Connor wrote:


I need to find a way to have the Polycom phones automatically park 
calls.  Right now my users hit #70# (I know the last # is optional 
but it speeds it up.) to park a call.  Personally I think this is 
easy, but my users would like one button to do this for them.  The 
Polycom has buttons we don't need (Transfer & Services), it would 
be nice if I could remap one of those buttons to dial #70#.  Or if 
I could add a soft button during a call that would work too.


Anyone have any suggestions on how to do this?

Thanks much,

Matthew O'Connor

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--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
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Re: [Asterisk-Users] detect SIP phone availability before dialing

[Paul Zimm]

Use application ChanIsAvail with the s option. This option only exists 
in CVS-HEAD version, the 1.0.x versions don't have this option.


from documentation:

If the option 's' is specified (state), will consider channel unavailable
when the channel is in use at all, even if it can take another call.


This is a pretty popular question.  IIRC SIP phones can't tell you 
their statuses, you need to send a call to them and determine whether 
or not they're Busy Now...


[EMAIL PROTECTED] wrote:


Hello,

 I need to detect availability of SIP phone before dialing. I need to
know if phone is  BUSY, CHANUNAVAIL before dialing. If phone is 
"free", then I will dial

it.




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Re: [Asterisk-Users] SIP behind NAT to pub Asterisk, best solution?

[chentschel]

Mensaje citado por: Blake Krone <[EMAIL PROTECTED]>:

> What is the best solution? I dont want to have modify firewall\'s at all or
> do port fowarding. Ideally I would like a solution that with either a
> softphone or wireless hardphone one could connect via friends, family, or
> hotspots without reconfiguring their devices.
>  What are people using? STUN? SER?
>  Thanks in advance!
>  -blake
> 
Give a try to the sip-helper for netfilter, and please let me know if this 
works for ya. 
Thanks. 
Christian. 
__
Registrate desde 
http://servicios.arnet.com.ar/registracion/registracion.asp?origenid=9 y 
participá de todos los beneficios del Portal Arnet.
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[Asterisk-Users] Feature codes work on SIP phone but not analog?

[Doug]

Hi,

This is what I have in "extensions_custom.conf":

; Time of Day functionality:
exten => *60,1,Answer
exten => *60,2,Wait(1)
exten => *60,3,SayUnixTime(,,IMSP)
exten => *60,4,Hangup

It works on a Cisco 7940 IP Phone, but on
analog phones, when I dial *60, I just
get a dial tone.  If I dial *60#, then
I just get a fast busy.

What's going on?

Also, how can I get Feature Codes to work
for all contexts?

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Re: [Asterisk-Users] Bulk Buys/Group Buys

[Ariel Batista]

Nathan Pralle wrote:

Hey folks,

Anyone know of companies selling bulk SIP adaptors (phones, adaptors,
etc.) or has the list ever considered doing something like a bulk buy?


Give a call to VoipSupply.com 800-398-VOIP (8647)


I was just curious...I'm looking to get another 5-6 Grandstreams or
similar and I figured I'd ask the list.  If we found something that
lots of people wanted, it probably couldn't hurt to contact a company
and ask for bulk deals.

Whadya think?  Anyone tried this before?

Nathan

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[Asterisk-Users] send Q931 information element keypadfacility ?!

[Bruno Voigt]

Hi all,

I'm looking for a way with any asterisk-version with TE410P (cpe
EuroISDN, Q931)
for sending an INFORMATION ELEMENT KeypadFacility,
eg. *87, during a connected call to the PSTN switch.

Are there existing functions in asterisk to generate & send such IE ?

If not what existing modules would be best to derive from?

TIA,
Bruno

begin:vcard
fn:Bruno Voigt
n:Voigt;Bruno
org:IC3S AG
adr:;;Baeckerbarg 6;Wilstedt;;D-22889;Germandy
email;internet:[EMAIL PROTECTED]
tel;work:+494109555105
tel;fax:+4941095
tel;cell:+4970068600686
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Re: [Asterisk-Users] Polycom: Button Remapping, HELP!

[Matthew T. O'Connor]

While I don't have it working yet, I think I have it figured out.  I 
have to add  entries to my sip.conf  Based on your example I was 
able to find the relevant info in the Polycom SIP 1.5 Admin Guide 
section 4.6.1.15.


My next question, which I haven't found in the admin guide (at least not 
yet) is where to you get a list of the buttons and their respective numbers?


Thanks again,

Matthew


Mojo with Horan & Company, LLC wrote:
Do you already have an  block in your sip.cfg?  add the 
 in there:

Try putting:

  ...
  ...
  key.IP_500.31.subPoint.prim="3"/>



Moj

Matthew T. O'Connor wrote:
Ok, that would be helpful for me with some other problems, however I 
don't see "I'm using the 1.5.2 Sip firmware the the conf files that came with 
that, so I don't have an ipmid.cfg file.  Is this something I can 
just add to my sip.conf?


Anyone out there any suggestions on how to do the speed dial "in-call"?

Thanks,

Matt



Mojo with Horan & Company, LLC wrote:

Matthew, when I tried this, I couldn't get the soundpoints to dial 
in-call.  They thought there were picking up a new line for a new call.


I created a speed-dial entry (in MACADDRESS-directory.xml, 
"Park#70#3") and then in 
ipmid.cfg:
key.IP_500.31.subPoint.prim="3"/>


This tells the phone to run Speed Dial 3 whenever the Services 
button (button #31 on a 500/501) is pressed.  I hope someone can 
help us configure them now to dial these digits in-call...


Mojo

Matthew T. O'Connor wrote:

I need to find a way to have the Polycom phones automatically park 
calls.  Right now my users hit #70# (I know the last # is optional 
but it speeds it up.) to park a call.  Personally I think this is 
easy, but my users would like one button to do this for them.  The 
Polycom has buttons we don't need (Transfer & Services), it would 
be nice if I could remap one of those buttons to dial #70#.  Or if 
I could add a soft button during a call that would work too.


Anyone have any suggestions on how to do this?

Thanks much,

Matthew O'Connor

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Re: [Asterisk-Users] Dial DTMF after bridging call

[Dinesh Nair]

On 10/12/05 15:41 Corey Frang said the following:
Interestingly, I started playing with the numbers on my phone after the 
dial messed up, and I could get the DTMF tones "stuck" playing one tone 
for a long time.  If i took the D() out of it It didn't have that problem.


On Aug 25, 2005, at 15:04, Joseph wrote:


Is there a way to dial DTMF after bridging the call.
The current option D() in Dial will dial DTMF before the call is bridged
and this doesn't do the job.
I need to dial DTMF after the call is bridged and the message is played
with "Background"


*CLI> show application senddtmf

  -= Info about application 'SendDTMF' =-

[Synopsis]
Sends arbitrary DTMF digits

[Description]
  SendDTMF(digits[|timeout_ms]): Sends DTMF digits on a channel.
  Accepted digits: 0-9, *#abcd
 Returns 0 on success or -1 on a hangup.

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

[Blake Krone]

What is the  best solution? I dont want to have modify firewall's at all or do port fowarding. Ideally I would like a solution that with either a softphone or wireless hardphone one could connect via friends, family, or hotspots without reconfiguring their devices.

 
What are people using? STUN? SER?
 
Thanks in advance!
 
-blake
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[Asterisk-Users] Bulk Buys/Group Buys

[Nathan Pralle]

Hey folks,

Anyone know of companies selling bulk SIP adaptors (phones, adaptors, 
etc.) or has the list ever considered doing something like a bulk buy?


I was just curious...I'm looking to get another 5-6 Grandstreams or 
similar and I figured I'd ask the list.  If we found something that lots 
of people wanted, it probably couldn't hurt to contact a company and ask 
for bulk deals.


Whadya think?  Anyone tried this before?

Nathan

--
-
Nathan E. Pralle
Give the Director a Serpent Deflector
www.nathanpralle.com
-
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Re: [Asterisk-Users] E400P vs te410p vs te411p

[MvPhone]

Hi,

Check out http://store.pbxhardware.com => it has better prices on the
E400P / T400P cards. There are also 2 port versions of these.

The difference between the TE4XX cards is there is no echo canceller
and the PCI chipset doesn't handle the master mode -> that eats a
little bit of CPU time.

regards
Martin

>Hi,
>
>I found E400P quad PRI card quite cheap (749USD):
>
>http://www.govarion.com/product_info.php?cPath=1&products_id=2&osCsid=68cdd6e3d08754
>
>in comparison to te410p (approx 1500 USD )
>
>http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TE410P
>
>Now newer generation with HW echo canceling emerged (te411p).
>
>I'm not sure in what things those two cards differ and what would be
>best
>option to buy (I believe there is big performance gap between them, but
>don't know how big and if it's worth of money) Also how do you find
>HW
>echo canceling in te411p ?
>
>Any advice, help ?
>
>Thanks in advance,
>
>regards,
>
>Rob.
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[Asterisk-Users] displaying a message on the Snom 320 using sipsak

[Franklin Webb]

Greetings fellow list members,
    It seems like a lot of 
people have been having trouble getting indicators working on the Snom 
phones, myself included.  Recently I was able to get the "desktop" 
functionality of sipsak to work on my Snom320, and I thought I would share what 
I could with the list.  For those not familiar this will replace the 
standard display when you are not on a call (normally showing the registered 
extension) with a text message of your choosing.  Our intent is to update 
this when our agents log into, and out of, queues.  This will give a visual 
indicator for agents and supervisors in our call center as to whether or not the 
phone is logged in, which is a large concern for us, and probably any call 
center.
 
For the record I tried this with a Snom360 also and 
could not get it working.
 
1.  Setup the phone in Asterisk as 
normal
2.  Get and install sipsak.  It can be 
found at http://sipsak.org/ (can be on any 
machine on your network, we used a Fedora Core 3 machine for this).
3.  In the Snom320 Configuration, under 
the "SIP" tab of your extensions line (Line 1 for me) make sure "Support 
Broken Registrar" is set to "on"
4.  In the Snom320 Configuration, 
under "Advanced" make sure "Filter Packets from Registrar" is set to 
"off"
5.  In the Snom320 Configuration, under 
"Advanced" under "Network identity (port):" set it to "5060" (you 
might be able to use a different port in here and in the sipsak command, 
but this is what worked for me.
6. Reboot the phone (just to be sure the settings 
take)
 
Then use the following sipsak command:
 
sipsak -vvv -M -O desktop -B "Test Msg" -r 5060 -s 
sip:[EMAIL PROTECTED]
 
where:
    "Test Msg" is the message you 
want displayed.  To turn the message off just set it to empty string 
("").
    5060 is the port, you could try 
another port here if you set your phone to another port under 
"Advanced"
    6670 is the extension of the 
phone
    192.168.51.251 is the IP of the 
PHONE, not the Asterisk server.  It does not appear that you can use the IP 
of the Asterisk server.
 
You can get a list of phones with IPs using the 
Asterisk command "sip show peers".  Our intent is to build a simple 
database matching extension to IP and then execute sipsak commands from a 
script, probably in the manager API, when agents log in and out that will 
update the phone display accordingly.
 
I hope this is helpful to some of you.
 
Franklin Webb
InterMedia Marketing Solutions 
 
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Re: [Asterisk-Users] detect SIP phone availability before dialing

[Mojo with Horan & Company, LLC]

This is a pretty popular question.  IIRC SIP phones can't tell you their 
statuses, you need to send a call to them and determine whether or not 
they're Busy Now...


[EMAIL PROTECTED] wrote:

Hello,

 I need to detect availability of SIP phone before dialing. I need to
know if phone is 
 BUSY, CHANUNAVAIL before dialing. If phone is "free", then I will dial

it.

 I need for automatic callback (.call files), but I need to know if it
is available both
 SIP phones before calling.

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--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
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Re: [Asterisk-Users] Polycom: Button Remapping, HELP!

[Mojo with Horan & Company, LLC]

Do you already have an  block in your sip.cfg?  add the 
 in there:

Try putting:

  ...
  ...
  key.IP_500.31.subPoint.prim="3"/>



Moj

Matthew T. O'Connor wrote:
Ok, that would be helpful for me with some other problems, however I 
don't see "using the 1.5.2 Sip firmware the the conf files that came with that, so 
I don't have an ipmid.cfg file.  Is this something I can just add to my 
sip.conf?


Anyone out there any suggestions on how to do the speed dial "in-call"?

Thanks,

Matt



Mojo with Horan & Company, LLC wrote:

Matthew, when I tried this, I couldn't get the soundpoints to dial 
in-call.  They thought there were picking up a new line for a new call.


I created a speed-dial entry (in MACADDRESS-directory.xml, 
"Park#70#3") and then in 
ipmid.cfg:
key.IP_500.31.subPoint.prim="3"/>


This tells the phone to run Speed Dial 3 whenever the Services button 
(button #31 on a 500/501) is pressed.  I hope someone can help us 
configure them now to dial these digits in-call...


Mojo

Matthew T. O'Connor wrote:

I need to find a way to have the Polycom phones automatically park 
calls.  Right now my users hit #70# (I know the last # is optional 
but it speeds it up.) to park a call.  Personally I think this is 
easy, but my users would like one button to do this for them.  The 
Polycom has buttons we don't need (Transfer & Services), it would be 
nice if I could remap one of those buttons to dial #70#.  Or if I 
could add a soft button during a call that would work too.


Anyone have any suggestions on how to do this?

Thanks much,

Matthew O'Connor

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--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
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Re: [Asterisk-Users] RE: faxing to/from asterisk - new scripts

[Thameem Ansari]

Please send them to my email  [EMAIL PROTECTED]

Thanks,
ThameemOn 10/12/05, Carlos Alperin <[EMAIL PROTECTED]> wrote:
















Can you send me those scripts to calperin
senecacom.net.?

 

Thanks in advance.

 

Carlos Alperin

Senior System Engineer

Seneca Communications, LLC

[EMAIL PROTECTED]


 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of 
Technical Support
Sent: Friday, October 07, 2005
10:55 AM
To: asterisk-users@lists.digium.com;
'Roman'
Subject: [Asterisk-Users] RE:
faxing to/from asterisk - new scripts



 

Roman: 



 





I created two bash scripts called Mail2Fax and Fax2Mail for
use with the asterisk sever.





 





They leverage the app_txfax and app_rxfax scripts, along
with ast_fax.  They make using these apps a lot easier, including being
able to mail to [EMAIL PROTECTED] for outgoing
faxes and then extracting phone numbers from the subject line!  (Makes it
easy to use with Sendmail without complex rules / virtual user tables).





 





They also include error logs, parameter checking, etc.





 





Let me know if you want them





 





Michelle Dupuis
Technical Support Specialist
Oxford Consulting Group Ltd.
Making IT work for your
business...



 





T: (519) 672-8238
E: 
[EMAIL PROTECTED]
W: 
www.ocg.ca
 











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Re: [Asterisk-Users] Polycom: Button Remapping, HELP!

[Matthew T. O'Connor]

Ok, that would be helpful for me with some other problems, however I 
don't see "using the 1.5.2 Sip firmware the the conf files that came with that, so 
I don't have an ipmid.cfg file.  Is this something I can just add to my 
sip.conf?


Anyone out there any suggestions on how to do the speed dial "in-call"?

Thanks,

Matt



Mojo with Horan & Company, LLC wrote:
Matthew, when I tried this, I couldn't get the soundpoints to dial 
in-call.  They thought there were picking up a new line for a new call.


I created a speed-dial entry (in MACADDRESS-directory.xml, 
"Park#70#3") and then in 
ipmid.cfg:
key.IP_500.31.subPoint.prim="3"/>


This tells the phone to run Speed Dial 3 whenever the Services button 
(button #31 on a 500/501) is pressed.  I hope someone can help us 
configure them now to dial these digits in-call...


Mojo

Matthew T. O'Connor wrote:
I need to find a way to have the Polycom phones automatically park 
calls.  Right now my users hit #70# (I know the last # is optional 
but it speeds it up.) to park a call.  Personally I think this is 
easy, but my users would like one button to do this for them.  The 
Polycom has buttons we don't need (Transfer & Services), it would be 
nice if I could remap one of those buttons to dial #70#.  Or if I 
could add a soft button during a call that would work too.


Anyone have any suggestions on how to do this?

Thanks much,

Matthew O'Connor

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RE: [Asterisk-Users] RE: faxing to/from asterisk - new scripts

[Carlos Alperin]

Can you send me those scripts to calperinsenecacom.net.?

 

Thanks in advance.

 

Carlos Alperin

Senior System Engineer

Seneca Communications, LLC

[EMAIL PROTECTED]

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Technical Support
Sent: Friday, October 07, 2005
10:55 AM
To: asterisk-users@lists.digium.com;
'Roman'
Subject: [Asterisk-Users] RE:
faxing to/from asterisk - new scripts



 

Roman: 



 





I created two bash scripts called Mail2Fax and Fax2Mail for
use with the asterisk sever.





 





They leverage the app_txfax and app_rxfax scripts, along
with ast_fax.  They make using these apps a lot easier, including being
able to mail to [EMAIL PROTECTED] for outgoing
faxes and then extracting phone numbers from the subject line!  (Makes it
easy to use with Sendmail without complex rules / virtual user tables).





 





They also include error logs, parameter checking, etc.





 





Let me know if you want them





 





Michelle Dupuis
Technical Support Specialist
Oxford Consulting Group Ltd.
Making IT work for your
business...



 





T: (519) 672-8238
E: [EMAIL PROTECTED]
W: www.ocg.ca 










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Re: [Asterisk-Users] MWI for endpoints not registered at Asterisk

[Ryan Hulsker]

Take a look at sipsak. http://sipsak.org/
Or at the wiki on how to use it.
http://www.voip-info.org/wiki-Asterisk+at+large

I think you will still need to be able to look up the IP address that
corresponds to your sip client though.

Ryan Hulsker


On Wed, 2005-10-12 at 08:21, Peter Bowyer wrote:
> On 12/10/05, Stojan Sljivic - Pamet <[EMAIL PROTECTED]> wrote:
> > Hi,
> >
> > We have phones registered at another soft switch, and would like to use
> > Asterisk as a Voicemail system.
> > Is it possible and how to configure Asterisk to send NOTIFY messages (for
> > MWI) to the endpoints that are not registered to the Asterisk?
> 
> I couldn't find a way round this, and ended up using a 'spare' line
> presentation on my GXP-2000 phones to register to the voicemail server
> simply to pick up the NOTIFYs. Since the phone only has a single MWI
> LED, it doesn't matter which line the NOTIFY comes in on.
> 
> Peter
> 
> --
> Peter Bowyer
> Email: [EMAIL PROTECTED]
> Tel: +44 1296 768003
> VoIP: sip:[EMAIL PROTECTED]
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[Asterisk-Users] ACD/queues question

[Pedro Nunes]

Hi there,

 

Does anyone know how to
setup an overflow queue? When a call rings on the queue A, if all agents were
busy, the call goes to the queue B.

If all agents in queue B were
busy, then the call stays on both queues until somebody answers it. 

 

I think this is a basic
ACD feature available on most PBX that support ACD functionality. 

Does anybody knows how to
do it with asterisk??

 

 

Thanks in advance

 

 

Pedro Nunes



 






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Re: [Asterisk-Users] Calibrating both RX and TX gain?

[Shaw Terwilliger]

On Wed, Oct 12, 2005 at 12:05:32PM -0400, [EMAIL PROTECTED] wrote:
> I am in the middle of trying to get a milliwatt test line to calibrate the 
> rxgain properly.  However, this won't help me with the txgain, will it? 
> How can I properly calibrate the txgain?  By ear?  Or is there a more 
> scientific method?

Maybe I can help.

I had a similar problem.  I was using a Digium TE205P card and two
Rhino channel banks, and every call that was bridged from a phone on
an FXS interface to a PSTN line on an FXO interface was (1) loud and
(2) had an echo with a tiny delay (maybe 30ms).  The echo sounded almost
like excess sidetone, but was delayed enough to phase shift the speech
and make things sound hollow.  I could verify that what was being
transmitted was coming back on the RX channel of the PSTN interface
(using ztmonitor).  I'm using Nortel analog, wall-powered phones (pretty
nice models).

I had echo cancellation on, and had tried all possible configuration 
settings for taps, etc.  Nothing killed my echo.

I had tried adjusting all the gains down in Asterisk for all the interfaces,
but that didn't work.

I contacted Rhino to see if they had any suggestions, and they were
able to give me a few.  What finally worked was setting the Asterisk gains
back to 0 for all channels, then adjusting the gains down on the channel banks
themselves for the phone (FXS) interfaces only.  A huge improvement!  My
current adjustements are the following:

On the Rhino channel banks:

  For FXS (phones) interfaces:

rx -4 dB
tx -4 dB

  For FXO (PSTN lines) interfaces:

rx 0 dB (default)
tx 0 dB (default)

In Asterisk's zaptel.conf:

  context=phones
  rxgain=3.0; This is to compensate for the drop in volume because of
; the -4 dB setting on the channel bank for rx.
  txgain=3.0; This is to compensate for the drop in volume because of
; the -4 dB setting on the channel bank for tx.

  context=pstn
  rxgain=1.4; This was bumped up last, as a result of a milliwatt test.
;
  txgain=1.4; This was also bumped up, because it makes the outbound
; calls a bit louder, and doesn't seem to overdrive the 
; line.  I figure the gain loss on rx (which was calibrated
; with the milliwatt test) should be similar to tx gain lost,
; although I couldn't directly test this.

Now, when I turn on echo cancellation for all my interfaces, the echo is
completely gone.  After compensating for the gain drop on the channel banks
with asterisk boost, the call volumes sound good too.

-- 
Shaw Terwilliger <[EMAIL PROTECTED]>
SourceGear LLC


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Re: [Asterisk-Users] Calibrating both RX and TX gain?

[Mojo with Horan & Company, LLC]

Hello :)

For example, once I have the rxgain calibrated for all of the lines, 
could I then call into, say, Zap/3 from Zap/4 and run Milliwatt() on 
Zap/3 and use ztmonitor on Zap/4 to calibrate it?  I'm sure it's not 
perfect, but would it be close enough?
That's exactly what you do.  Once I had adjusted my rxgains to calibrate 
them to the signal that the phoneco gave me, I just dialed out of one 
line and into another.  Everything's supposedly digital on the phoneco 
side, so no loss should occur.  (because with the rxgains you've already 
compensated for what will happen on the inbound trip through the 
copper).  So by then adjusting your txgains on each channel, you can 
feel confident that the phoneco is accurately representing to you how 
you sound from its point of view.


A second question:  doesn't it seem wrong that my rxgain and txgain are 
so far off when I'm just talking to a channel bank 12 feet away?  I sure 
don't have cable loss.  It sure seems like the impedance is way off or 
something.  Is there a way to test this further, rather than just 
cranking up the gain?  My guess is that using the milliwatt line will 
just tell me to make the rxgain higher, which will probably just make 
the echo issues worse...
It does seem like something else is wrong.  You shouldn't require such 
high rxgains in my opinion, but I have no idea what could be causing 
this need.


Mojo
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RE: [Asterisk-Users] Real Life FAX sending receiving

[Jenna Cole]

thanx, i did it.
i just instaled the debian packages for asterisk and
asterisk-app-fax and its working this way

fax--fxs--ipnetwork--fxs--fax

obviously the fxs ports are digium cards in linux
machines running asterisk using sip. the fax quality
is perfect

now i am trying with the following scenario:

fax--fxs--ipnetwork--FXO--pbx--fax

when i transmit a fax, an 80 percent of the fax is OK
but there are lines that cannot be read because the
quality is bad. sometimes the lines overlap each
other, and sometimes the height of the line is smaller

Can anyone help me?

Thanx 

 --- [EMAIL PROTECTED] escribió:

> I can send/receive just fine on an eicon bri to a
> zaptel analog
> interface.
> 
> I would say, if you wish to use faxing on a regular
> basis to a remote
> proxy though, you're possibly better off with a
> landline.
> 
> Regards,
> Greg 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On
> Behalf Of Doug Lytle
> Sent: Monday, October 03, 2005 3:23 PM
> To: Asterisk Users Mailing List - Non-Commercial
> Discussion
> Subject: Re: [Asterisk-Users] Real Life FAX sending
> receiving
> 
> Jenna Cole wrote:
> 
> >receive the fax via SIP and send it to my
> faxmachine.
> >I also want to send a fax from my faxmachine
> through the digium card, 
> >so asterisk should send the fax via SIP to the
> gateway, which also has 
> >a faxmachine connected.
> >
> >is this possible?
> >  
> >
> Short answer, no.  Long answer can be found here:
> 
> http://www.soft-switch.org/spandsp_faq/ar01s04.html
> 
> Doug
> 
> -- 
>  
> Ben Franklin quote:
> 
> "Those who give up essential liberties for temporary
> safety deserve
> neither liberty nor safety."
> 
> 
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RE: [Asterisk-Users] AGI and set_callerid for number and name

[Pedro Nunes]

Curse,

Look at this php script ...

Contactlookup.agi

#!/usr/local/bin/php -q
  4501,1,agi,contactlookup.agi
exten => 4501,2,SetCIDName(${Name})
exten => 4501,3,Dial(SIP/421,15)


It looks to an mssql DB, try to find the callerID number in table
"extensions", and then sets a variable named "Name" to the value of
table "Name". Cool hah...



Pedro Nunes



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Serge
Lhermitte
Sent: quarta-feira, 12 de Outubro de 2005 17:57
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] AGI and set_callerid for number and name


Hi,


I've been trying to use the set_callerid function in the AGI. It sets
the CallerIDname properly but I can't figure out how to set the
CallerIDnumber. 

Is it at at possible ?

Cheers.
SL


 
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[Asterisk-Users] sound very loud (saturated) through IAX2 and SIP

[Goran]

I have very loud sound through IAX2 and SIP channels, even very
saturated in some moments.

Why? How to change sound level (on IAX2 and SIP channels)?

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[Asterisk-Users] AGI and set_callerid for number and name

[Serge Lhermitte]

Hi,


I've been trying to use the set_callerid function in the AGI. It sets
the CallerIDname properly but I can't figure out how to set the
CallerIDnumber. 

Is it at at possible ?

Cheers.
SL


 
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