Boris Bakchiev | 1 Apr 01:08 2006
Picon

RE: Iaxmodem speed limit?

I Guess you can edit the following line in your hylafax config file for
your iaxmodems.

Class1RMQueryCmd:       "!24,48,72,96"

Put exclamation in front of 96 (as it is done with 24) and it should
disable the receive with that speed.

> 
> 	Is there a way to limit the speed of Hylafax to 7200 bits/s to
get
> more
> reliable results?

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Michael Wallette | 1 Apr 01:49 2006

Re: Asterisk-Users Digest, Vol 20, Issue 226

That was how I reset the black Iaxy I have used; I've never used a blue one.

What I found was the initial provisioning would work fine, but if I
tried to change the settings after having already provisioned the
device, the provisioning program would hang, so I Googled for
instructions on resetting the Iaxy to the factory settings. This was the
procedure described at http://www.voip-info.org/wiki-IAXy . It *may* not
be necessary to go through all of these steps to reset it, but it worked
for me. YMMV...

After provisioning, the Iaxy works great; I'd recommend it to anyone
looking for an inexpensive, simple FXS.

--Mike

>Message: 9
>Date: Fri, 31 Mar 2006 09:05:45 -0900
>From: "Mojo with Horan & Company, LLC" <mojo <at> horanappraisals.com>
>Subject: Re: [Asterisk-Users] IAXY codec support and questions..
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>	<asterisk-users <at> lists.digium.com>
>Message-ID: <442D6F79.7010905 <at> horanappraisals.com>
>Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
>Really all that?  In my experience, which is admittedly only on the blue 
>iaxys and not the black ones, was just provision it, unplug power, 
>replug power.  The black one I have now I haven't had to re-provision 
>yet so this could be so on them....
>
>
(Continue reading)

James Harper | 1 Apr 01:56 2006
Picon

RE: Re: BRI cards, HFC, and bristuff - a general questionto clear up my understanding.

> Appreciate the replies everyone -- really
> 
> I'm wondering if I should be using zapHFC with my Junghanns card
instead
> of
> qozap?  Everyone always mentions zaphfc -- mostly I guessed because
they
> are
> using a zaphfc-compatible card ----- but *maybe* I should try that
instead
> of qozap???
> 
> And yep -- totally know about the module load order thing and ztcfg --
no
> worries there
> 
> I've been able to dial out and everything from the start ! -- which is
a
> bridge from digium-->junghanns there......but incoming calls seem to
be a
> whole other issue..... :-(
> 
> 
> Exhausted from trying a million things,
> 

I haven't been following this thread at all, so forgive me if I'm
suggesting stuff that you've already been told a billion times, but
isn't the qozap driver only for the quad/octo cards?

(Continue reading)

John Fulton | 1 Apr 02:33 2006
Picon

Mitel 3300 PRI problems

We are attempting to interface Asterisk with a Mitel 3300.

Initially we tried FXO Ground start T1 and had many problems with that. Even spend 5 hours on the phone with a Digium engineer and were unable to get it working right.

Next we turned it up as an E&M T1 and that actually worked great. No problems at all.

Now we are testing a T1 PRI configuration and have been unable to get it to work right. We tried different switchtype configurations. We are back to "national" (NI-2). I am setup as pri_net. And I know that is working right because if I set the PRI for pri_cpe, I get error messages in the log that the remote end thinks it's a cpe and to change the signalling.

When a call it placed to the PRI, I get nothing. No answer, no data across the D channel, yet the Mitel says it is successfully grabbing a trunk in it's debugs.

I turned up pri intense debug span and the ONLY thing I get is the following over and over again. It seems to be at regular intervals and I see nothing else even when I place a call from the Mitel to the Asterisk server. Any ideas? Here's the debug output:

T203 counter expired, sending RR and scheduling T203 again
Sending Receiver Ready (115)

> [ 02 01 01 e7 ]

> Supervisory frame:
> SAPI: 00  C/R: 1 EA: 0
>  TEI: 000        EA: 1
> Zero: 0     S: 0 01: 1  [ RR (receive ready) ]
> N(R): 115 P/F: 1
> 0 bytes of data
-- Restarting T203 counter

< [ 02 01 01 e7 ]

< Supervisory frame:
< SAPI: 00  C/R: 1 EA: 0
<  TEI: 000        EA: 1
< Zero: 0     S: 0 01: 1  [ RR (receive ready) ]
< N(R): 115 P/F: 1
< 0 bytes of data
-- ACKing all packets from 114 to (but not including) 115
-- Since there was nothing left, stopping T200 counter
-- Stopping T203 counter since we got an ACK
-- Nothing left, starting T203 counter
-- Got RR response to our frame
-- Restarting T203 counter



---------------- http://www.telalaska.com -----------------
TelAlaska, Inc.                                 John Fulton
201 E 56th Ave.                    Senior Network Architect
Anchorage, Alaska                 Cisco Certified CCNA/CCDA
99518                               Microsoft Certified MCP
907 563 2003                        A+ Certified Technician
Fax 550 1646                             jfulton <at> arctic.net
------------------ http://www.arctic.net ------------------
Interior Telephone Company         Mukluk Telephone Company
Eyecom Cable                        TelAlaska Long Distance
TelAlaska Networks                Seward Net and Arctic Net
-----------------------------------------------------------
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sam | 1 Apr 02:49 2006
Picon

Re: Building Asterisk embedded device

mustardman29 wrote:

>Not a lot to go on sam.  What do you want to do?  If you just want to play
>or have very minimal requirements then get a soekris NET4801 board, CF and
>install Astlinux.
>http://www.soekris.com/net4801.htm
> 
>
>  
>
>>-----Original Message-----
>>From: sam [mailto:samwun <at> telpacific.com.au] 
>>Sent: Friday, March 31, 2006 6:01 AM
>>To: asterisk-users <at> lists.digium.com
>>Subject: [Asterisk-Users] Building Asterisk embedded device
>>
>>Hi,
>>
>>I want to build a PBX base on Asterisk using an embedded device.
>>Can anyone please recommend an embedded device I can use for doing so?
>>I will install linux or freebsd in the device.
>>
>>Thanks
>>A
>>
>>
>>    
>>
>Hi, is this hardware similar to Linksys pap5?
>  
>

Thanks
Sam

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Jon Webster | 1 Apr 02:56 2006

RE: Marketing Materials

Perhaps you would be willing to share it with the rest of the community?

-----Original Message-----
From: asterisk-users-bounces <at> lists.digium.com
[mailto:asterisk-users-bounces <at> lists.digium.com] On Behalf Of Dovid Bender
Sent: Friday, March 31, 2006 7:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Marketing Materials

We created out own. You should do the same.

--- Bob McDowell <bmcdowell <at> federalprotection.com>
wrote:

> 
> The owner of my company just asked me for an
> Asterisk brochure.  Has
> anyone seen such a creature?  I know of some really
> informative
> websites, but I think a pdf would be priceless at
> this point.
> 
> 
> Thanks,
> 
> Bob McDowell
> 
> 
> 
> 
> EMAIL PRIVELEGED & CONFIDENTIAL CLIENT COMMUNICATION
> 
> 
>    *** PRIVILEGED AND CONFIDENTIAL CLIENT
> COMMUNICATION ***
> This e-mail message and all attachments, if any, may
> contain confidential and privileged material and are
> intended only for the intended recipient.  Any
> unauthorized review, use, disclosure or distribution
> is prohibited.  If you are not the intended
> recipient, please contact the sender by reply e-mail
> or by calling  (417) 869-9192 and destroy the
> original and any copies of this e-mail.
> 
> 
> EMAIL PRIVELGED & CONFIDENTIAL CLIENT
> COMMUNICATION.DOCDKH
>  
>  
> 
> 
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> To UNSUBSCRIBE or update options visit:
>   
>
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Nicolás Gudiño | 1 Apr 02:49 2006
Picon

Re: Reporting?

> NICE!!!!!!!!!!!!!!!!!!!!!!!!!
>
> On 3/30/06, Joe Dennick <joe <at> dennick.net> wrote:
> > I see (and like) the demo, but where can we get it?
> >
> > Doug Lytle wrote:
> >
> > > Nicolás Gudiño wrote:
> > >
> > >> <shameless plug> Something like this perhaps?
> > >>
> > >> http://www.asternic.org/stats/demo

It is not released yet... I'm not having much time to write the web
page, documentation, tarball, etc.

--
Nicolás Gudiño
Buenos Aires - Argentina
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asterisk | 1 Apr 03:02 2006
Picon

Re: Re: How is Teliax ?

it says that there's no packet loss and the average latency to teliax is 
1/3 that of junction networks.

the traceroutes did on several occasions suggest that teliax's upstream 
provider in colorado (rockynet) was suffering from bandwidth overload, as 
the last hop from rockynet to teliax would jump around sometimes.

this still does not explain why teliax origination did not suffer from any 
problems. only teliax termination suffered.

yet both termination and origination on junction networks has no problems, 
despite 3x higher latency.

and i'm not the only one who experienced these exact problems.

so i use teliax for origination, and junction networks for termination.

-Dan

On Fri, 31 Mar 2006, Michael Welter wrote:

> And, what does traceroute say about your connection with Teliax?  Hmm?
>
> asterisk <at> anime.net wrote:
>> On Fri, 31 Mar 2006, Michael Welter wrote:
>>> Having said all that, I see where Teliax have installed the voip-co4 host 
>>> on Viawest.  Are you using that host for your analysis?
>> 
>> I have used every single gateway teliax has made available to me, including 
>> their beta test ones. I experienced choppiness with all of them, but only 
>> when terminating calls. As stated before, origination is perfect.
>> 
>> With junction networks -- no problems whatsoever with either termination or 
>> origination.
>> 
>> -Dan
>> _______________________________________________
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>> 
>> 
>
> -- 
> Michael Welter
> Telecom Matters Corp.
> Denver, Colorado US
> +1.303.414.4980
> mike <at> TelecomMatters.net
> www.TelecomMatters.net
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Josué Conti | 1 Apr 03:36 2006
Picon

Zap channels - help

I am installing one asterisk, to establish connection with my PABX Siemens, in ISDN, link went up normally, also I obtain to internally call the branches the PABX, normally, but when I try to dial for the PSTN, through pabx with the command                  exten = _ 19xxxxxxxx, 1, dial(zap/g2/${EXTEN}, 30) asterisk, reports me the following error:  
-- Executing Dial("SIP/8110-a729", "zap/g2/1971411234|30") in new stack
    -- Called g2/1971411234
    -- Channel 0/1, span 2 got hangup
    -- Hungup 'Zap/32-1'
  == No one is available to answer at this time                                                        
 However, when use the rule exten = _ 7xxx, 1, dial(zap/g2/${EXTEN}, 30) I obtain to call the branches pabx, normally.
-- Executing Dial("SIP/8110-71ee", "zap/g2/7500|30") in new stack
    -- Called g2/7500
    -- Zap/32-1 is ringing
    -- Zap/32-1 answered SIP/8110-71ee
    -- Channel 0/1, span 2 got hangup
    -- Hungup 'Zap/32-1'
  == Spawn extension (default, 7500, 1) exited non-zero on 'SIP/8110-71ee'
Somebody would have some idea to help in this case me?
Greatings
Josué
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Richard OSS | 1 Apr 03:50 2006

Re: Mitel 3300 PRI problems

I had this same problem with SX2000. I think you have to configure the Mitel to "Speed Dial" to your Asterisk server. What I think is happening is that the PBX grabs a trunk but does not "dial" into Asterisk.
 
Ask the person configuring the Mitel PBX to setup an extension (e.g. 1234) when callers connect to the PBX. That's what they had to do to get data to my Asterisk.
 
Goodluck.
 
richard

John Fulton <jfulton <at> arctic.net> wrote:
We are attempting to interface Asterisk with a Mitel 3300.

Initially we tried FXO Ground start T1 and had many problems with that. Even spend 5 hours on the phone with a Digium engineer and were unable to get it working right.

Next we turned it up as an E&M T1 and that actually worked great. No problems at all.

Now we are testing a T1 PRI configuration and have been unable to get it to work right. We tried different switchtype configurations. We are back to "national" (NI-2). I am setup as pri_net. And I know that is working right because if I set the PRI for pri_cpe, I get error messages in the log that the remote end thinks it's a cpe and to change the signalling.

When a call it placed to the PRI, I get nothing. No answer, no data across the D channel, yet the Mitel says it is successfully grabbing a trunk in it's debugs.

I turned up pri intense debug span and the ONLY thing I get is the following over and over again. It seems to be at regular intervals and I see nothing else even when I place a call from the Mitel to the Asterisk server. Any ideas? Here's the debug output:

T203 counter expired, sending RR and scheduling T203 again
Sending Receiver Ready (115)

> [ 02 01 01 e7 ]

> Supervisory frame:
> SAPI: 00  C/R: 1 EA: 0
>  TEI: 000        EA: 1
> Zero: 0     S: 0 01: 1  [ RR (receive ready) ]
> N(R): 115 P/F: 1
> 0 bytes of data
-- Restarting T203 counter

< [ 02 01 01 e7 ]

< Supervisory frame:
< SAPI: 00  C/R: 1 EA: 0
<  TEI: 000        EA: 1
< Zero: 0     S: 0 01: 1  [ RR (receive ready) ]
< N(R): 115 P/F: 1
< 0 bytes of data
-- ACKing all packets from 114 to (but not including) 115
-- Since there was nothing left, stopping T200 counter
-- Stopping T203 counter since we got an ACK
-- Nothing left, starting T203 counter
-- Got RR response to our frame
-- Restarting T203 counter



---------------- http://www.telalaska.com -----------------
TelAlaska, Inc.                                 John Fulton
201 E 56th Ave.                    Senior Network Architect
Anchorage, Alaska                 Cisco Certified CCNA/CCDA
99518                               Microsoft Certified MCP
907 563 2003                        A+ Certified Technician
Fax 550 1646                             jfulton <at> arctic.net
------------------ http://www.arctic.net ------------------
Interior Telephone Company         Mukluk Telephone Company
Eyecom Cable                        TelAlaska Long Distance
TelAlaska Networks                Seward Net and Arctic Net
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Gmane