Re: [asterisk-users] Panasonic PBX connect to Asterisk

[C F]

Use a sip to PRI gateway or a PRI card in the asterisk system. Connect that
to the Panasonic TDA600 using a PRI card on the panasonic side (KX-TDA0290).
this will be the most worry free solution.

On Wed, Sep 14, 2016 at 2:05 AM, Harry McGregor 
wrote:

> Hi,
>
>
> You need to find out more about the configuration of this specific TDA600,
> as it could be either POTS or E1, once you know that, you can determine
> what options are best.
>
> -Harry
>
> On 09/13/2016 10:51 PM, Ikka Tirtawidjaja wrote:
>
> Dear Harry,
>
> Thx for the explanation.
>
> My team manage building's PBX that use Asterisk 13.x.
> We use Asterisk PBX for this buildings that have apartment and office
> customer.
> From my Asterisk PBX, we connect to IPPhone (yealink) or ATA Converter
> (cisco SPA112).
> Others are using PBX like panasonic analog, audiocodes SBC, etc, and we
> use ATA Converter to convert from SIP to Analog (CO Line)
>
> Now, we have a new customer (tenant) that have Panasonic TDA600.
> If we use FXS or ATA Converter, its going to have a lot of that, because
> this tenant going to use about 60 ext / sip line.
> Replacing asterisk PBX on my (company) side or replace TDA600 on my
> customer side is not acceptable.
> So we need to find a "win-win" solution for this.
>
> Thx in advance,
>
>
> Ikka
>
>
>
>
> On Wed, Sep 14, 2016 at 12:40 PM, Harry McGregor 
> wrote:
>
>> Hi,
>>
>> On 09/13/2016 06:51 AM, Ikka Tirtawidjaja wrote:
>>
>> Hi,
>>
>> Is there anyone here who has experience connecting Asterisk (ver 13.8)
>> with PBX Panasonic KX-TDA600 ?
>>
>> The architecture more less like this :
>>
>>
>> Telco Sip Trunk ---> Asterisk 13 ---> Panasonic KX-TDA600 ---> Phone / Fax
>>
>>
>> What connectivity do you currently use for the KX-TDA600?  E1, T1, POTS,
>> BRI?
>>
>> Others have suggested a SIP to E1/T1 gateway, which would let you skip
>> the asterisk box, if you don't have other uses for it.
>>
>> Another option is to use a PCI-E E1/T1 interface card in the asterisk
>> box, especially if you already have an E1 or T1 interface in the KX-TDA600.
>> I personally don't like buying smaller then a dual T1/E1 card, as the price
>> difference between a dual and a single is so small.  If the KX-TDA600 is
>> set-up for Analog/POTS, you can use a channel bank on the second T1/E1
>> port, and feed POTS into the KX-TDA600.
>>
>> For a small installation that wanted to keep their Nortel Key System, and
>> their Telco really wanted to provide a PRI instead of POTS (the Nortel
>> could only take pots), we used a dual T1 PCI card in an asterisk box, ran
>> PRI on the Telco interface, an ADIT 600 channel bank on the second
>> interface, and handed 4 POTS lines to the Nortel Key System.
>>
>> The key is to give your self the most flexibility to change later, and
>> preserve your existing investment.
>>
>> -Harry
>>
>> Thanks in advance,
>>
>>
>> Regards,
>>
>> Ikka - Jakarta, Indonesia
>>
>>
>>
>>
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>
>
>
>
>
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>
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Re: [asterisk-users] What linux distro most popular for Asterisk

[C F]

Slackware here.


On Thu, Oct 17, 2013 at 8:57 PM, Tiago Geada tiago.ge...@gmail.com wrote:

 debian wheezy compiled asterisk from source


 On 18 October 2013 00:27, Andrew Furey andrew.fu...@gmail.com wrote:

 [Apologies, top-posting, Gmail, yadda yadda]

 As with a lot of software, I suspect the best answer is whichever distro
 YOU are most comfortable with. You're the one who has to support it, after
 all... Just my 2c.

 Andrew


 On Thursday, 17 October 2013, Rusty Newton wrote:

 On Tue, Oct 15, 2013 at 11:58 PM, Michelle Dupuis mdup...@ocg.ca
 wrote:
  Is there a recent survey of that Linux distro and version people are
 using
  for the Asterisk installations?  I recall seeing a pie chart over a
 year ago
  (I think on a wiki but I can't find it again)also hoping for
 something
  more current.
 
  I suspect RH5 and RH6 are most popular...but I'm looking for facts

 I don't have any numbers, but I watch the issue tracker a lot and I
 see pretty much CentOS, Debian and Ubuntu. Which seems to match what
 everyone else is saying on this thread.

 --
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 Digium, Inc. | Community Support Manager
 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
 direct: +1 256 428 6200

 Check us out at: http://digium.com  http://asterisk.org

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Re: [asterisk-users] OT - Multi Function Printer with one touch scanning/emailing

[C F]

There are some appliances that support it is well. But those don't
have a scanner, just thru the computer. MultiTech FaxFinder comes to
mind, for the price they are excellent.

On Fri, Jul 6, 2012 at 3:13 AM, Olivier oza_4...@yahoo.fr wrote:
 2012/7/5, C F shma...@gmail.com:
 I searched a bit more,
 http://www.muratec.com/catalog/F320_config.html#email
 The above model supports t.37

 That's very interesting to know.
 I quickly googled for t.37 and found several other vendors mentioning
 this (some from rather old documents).
 The strange thing is some vendors seem to completely ignore t.37 and
 t.38 (google for hp t.37 or hp t.38 ).

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Re: [asterisk-users] OT - Multi Function Printer with one touch scanning/emailing

[C F]

T.37
http://en.wikipedia.org/wiki/T.37_(ITU-T_recommendation)
There were some scanners manufactured with this in mind, however I
cant remember who made them.

On Thu, Jul 5, 2012 at 10:15 AM, Olivier oza_4...@yahoo.fr wrote:
 Hi,

 I'm curious about the availability of Multi Function Printers with the
 following feature :
 - user feeds paper sheets in
 - user dials a phone number (0123456, for instance) then a hits single button
 - the result is that the paper sheets are scanned into a file which is
 emailed to a given address such as 0123...@myfaxgateway.com (where
 myfaxgateway.com is a fixed and configured address).

 Is this a common feature ?
 Last time I checked, MFP's alphanumeric diaplan was either oriented to
 digits or letters typing, and of course, scanning feature implied
 letters typing mode.

 Regards

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Re: [asterisk-users] OT - Multi Function Printer with one touch scanning/emailing

[C F]

I searched a bit more,
http://www.muratec.com/catalog/F320_config.html#email
The above model supports t.37 but no sure if you can have it function
such that any number entered will actually be send to a gateway.

On Thu, Jul 5, 2012 at 10:20 AM, C F shma...@gmail.com wrote:
 T.37
 http://en.wikipedia.org/wiki/T.37_(ITU-T_recommendation)
 There were some scanners manufactured with this in mind, however I
 cant remember who made them.

 On Thu, Jul 5, 2012 at 10:15 AM, Olivier oza_4...@yahoo.fr wrote:
 Hi,

 I'm curious about the availability of Multi Function Printers with the
 following feature :
 - user feeds paper sheets in
 - user dials a phone number (0123456, for instance) then a hits single button
 - the result is that the paper sheets are scanned into a file which is
 emailed to a given address such as 0123...@myfaxgateway.com (where
 myfaxgateway.com is a fixed and configured address).

 Is this a common feature ?
 Last time I checked, MFP's alphanumeric diaplan was either oriented to
 digits or letters typing, and of course, scanning feature implied
 letters typing mode.

 Regards

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Re: [asterisk-users] Advertising oportunity.

[C F]

asterisk biz

On Tue, Jun 12, 2012 at 3:10 PM, Jonson Player jonsonpla...@gmail.com wrote:
 Hello,

 I don't know if this list is appropriated to this subject but I want to ask
 you if there's some list where I can make an advertising announce for a new
 sip web site that was just launched.
 hank you.


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Re: [asterisk-users] Free calls to the uS question

[C F]

Doesnt google voice offer that?

On Fri, Mar 30, 2012 at 1:00 PM, Christian christia...@runbox.com wrote:
 Hi all,
 Does anyone know of any providor that offers free calling to the US?
 Feel free to contact me off list.
 Many thanks,
 Christian

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Re: [asterisk-users] Polycom firmware 4.0.1 and paging

[C F]

Are you using the same cfg files?
If yes I would try rewriting them from scratch using the blank cfg files
that come with new firmware. I have seen wiered things by using olde cfg
files



On Friday, February 10, 2012, Mike l...@net-wall.com wrote:
 Hi,



 I just moved many Polycom phones from firmware v3 to 4.0.1b.  Anto-Answer
simply stopped functioning. I can downgrade and make it work, upgrading
kills it again. There obviously is a difference in how the newer firmware
is treating this auto answer sip header.



 Can anybody tell me if they have Polycom firmware 4.x.x working with
auto-answer/paging? Just so I know it’s worth my time to investigate, as
opposed to knowing it`s a Polycom firmware bug? If so, did you have to make
any changes to the SIP header sent to make Polycom phones auto answer?



 Regards,



 Mike












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Re: [asterisk-users] Is this doable?

[C F]

On Wednesday, February 8, 2012, Josh mojo1...@privatedemail.net wrote:

 http://www.asterisk.org/astdocs/node66.html

 Thanks, never knew that!

 Yes, I understand that it's not what you want, but that doesn't make it
a security concern.  If Asterisk is publicly available on one interface,
making it available on another interface doesn't make you less secure.

 You lost me. What I want/don't want is largely irrelevant. The issue is,
as you rightly pointed out, whether it is considered more secure or less
secure when Asterisk binds to 0.0.0.0 as oppose to using a specific set of
interfaces, selected at startup.

I don't get this. Didnt EVERYONE know it's insecure?


 If one has internal networks, accessible via, say eth1 and tun0, and
implements Asterisk to act as the internal/private PBX (without exposing it
to the outside world), then having been forced to use 0.0.0.0 will, of
course, expose Asterisk to any other - undesirable - interfaces, including
those pointing to the outside world.

 By having the option to specify which interfaces Asterisk should use to
bind to (via multiple {udp,tcp}bind statements or by any other means)
Asterisk is *not* exposed to any undesirable interfaces and thus, the risk
is not there. I thought I have made that clear by now, obviously I haven't,
it seems.

 It's fine if you want to take that step, but please drop the everyone
knows this is a security risk thing.  You appear to be alone in that
opinion, and unable to explain why you think it's a security risk.
Moreover, you're speaking for others without warrant or welcome.

 If you can't see why binding to 0.0.0.0 carries greater risk than
restricting Asterisk which interfaces to use, then you are truly blind and
beyond help, I am afraid.


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Re: [asterisk-users] Is this doable?

[C F]

On Wednesday, February 8, 2012, Josh mojo1...@privatedemail.net wrote:

 I don't get this. Didnt EVERYONE know it's insecure?

 Can you read?

Can everyone?



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Re: [asterisk-users] Router that support Asterisk

[C F]

G
Have you ever heard of Google?
Here is a link on google:
http://lmgtfy.com/?q=google


On Wed, Feb 1, 2012 at 2:17 PM, bilal ghayyad bilmar...@yahoo.com wrote:

 Hi All;

 I heard from some friends that there are a dsl router that has Linux OS
 and it has asterisk on it, so the ip phone can register on this router,
 also if the router has FXS or FXO ports then it can be used to place calls
 through them.

 Is it really? Where I can these routers? Did anyone try it to tell us if
 it is stable and working fine?

 Regards
 Bilal

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Re: [asterisk-users] Is this doable?

[C F]

Whats asterick?

On Wed, Feb 1, 2012 at 7:48 PM, Josh mojo1...@privatedemail.net wrote:
 I am trying to configure Asterick, having the following system setup on
 the Asterick server:

 * eth0 faces the external Internet interface, *but* it does not have IP
 address (it has a private one given to it by my ISP's DHCP server);
 * eth1 faces my internal network (say 10.1.1.0/24);
 * tun0 serves all mobile smartphones and connects to the internal
 network (it has a different ip range, say 10.1.2.0/24) - they are all
 connected via the Internet using OpenVPN;

 I would like to configure Asterick for internal calls between ourselves
 (eth1-tun0) and I think I have no problem with configuring this part.
 I would also like to use one external VOIP provider to which Asterick
 registers on startup. I think I know how to do that and use the
 register option in sip.conf, though I am not sure for the rest of the
 NAT-related entries (see below).

 The purpose of registering this external account is so that both the
 smart phones (tun0) and the internal net (eth1) users could use this
 account to make external calls (starting with 0, i.e _0[0-9].
 pattern in extensioins.conf). Obviously, I need these calls to be routed
 properly via the external VOIP account. In addition to that, I would
 also need to receive calls from that external account to a nominated
 internal one (say on extension 20).

 Is this achievable?

 If so, I am not completely clear on whether I need to explicitly specify
 my public IP address (via externip/externhost) or whether Asterick is
 able to find it without this option? If not, then my plan is to use
 external program to find it and then use a script in Asterick to set it
 up as an environment variable. Would that work? That external IP address
 is going to change, but only in rare circumstances and in such cases I
 have to restart a lot of stuff (including Asterick) on that server (this
 is usually triggered by a monitoring program), so it won't be a problem
 once it is setup initially. I am also not sure whether to specify
 nat=yes or just have nat=route only - any ideas?

 Is there a comprehensive list of all the options available in sip.conf
 and what they do, because I was unable to find such a list?

 If the above is doable, I would also like to add the following 2 features:

 1. Secondary external VOIP account, though I have no idea how to specify
 its port in register (it uses port 5065 instead of the standard 5060).
 That account would need to be used on a separate interface (eth2) with a
 different public IP address. Would it be possible to use
 externip/externhost inside that external account section to specify it?
 If this is not possible, then I am thinking of running a separate
 instance of Asterick with the second VOIP account/public IP address set
 up - would that work?

 2. I would like to be able to configure the following work flow: for a
 specific set of (external) calling numbers (including where no Caller ID
 is available):
 a) these callers to be prompted to specify the reason for their call;
 b) their response to be temporarily recorded/stored (a short message
 of, say no more than 10 seconds long or when they press '#' for that
 recording to stop);
 c) Asterick then rings the nominated number for external VOIP calls
 (extension 20) and play that recorded message back;
 d) then asks for one of four possible outcomes:
 - accept this call (pressing, say 1) in which case the call is connected
 as normal;
 - reject it with a message that that number/person is unavailable
 (say, by pressing 0);
 - ask the caller to leave a message by transferring them to a voicemail
 (say by pressing 2); or
 - end the initial call completely with a message that the caller/number
 has been blacklisted (say, by pressing the 9 key);

 Could this be achieved?

 One final question about binding: in order to be able to use both tun0
 and eth1 interfaces so that Asterick serves the calls from both eth1 and
 tun0, do I have to use bind 0.0.0.0? Is there an alternative, like
 specifying bind 10.1.1.1 for eth1 and then bind 10.1.2.1 for the
 tun0 interface - is this possible?

 Many thanks in advance!


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Re: [asterisk-users] externip nat audio sip trunk issue problem

[C F]

On Wed, Feb 1, 2012 at 9:14 PM, Gabriel Ortiz Lour
ortiz.ad...@gmail.com wrote:
 Hi all,

   I've tried search this problem on the list... no luck...

   The case is:

 without externip/localnet config on sip.conf [general] my SIP trunk works,
 but with no audio NAT problem (asterisk sends the private 192 address to the
 outside...)

 when I configure externip/localnet correctly my SIP trunk simply disappear!
 Checking the signalling with tcpdump shows me that Im sending the packets to
 the correct SIP trunk IP but there is no response AT ALL from it...

Can you explain this?
What do you mean no response? Is it registering? Do you have a debug output?


 Anyone had this problem?

 Thanks,
 Gabriel

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Re: [asterisk-users] Cell Phone as a Queue member

[C F]

Use local channel

2012/1/31 Niccolò Belli darkbas...@gmail.com:
 Hi,
 Is it possible? I tried AddQueueMember(SIP/$TRUNK/number) but it tries to
 call SIP/$TRUNK instead.

 Cheers,
 Darkbasic

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Re: [asterisk-users] Real T1 trunk group...

[C F]

On Mon, Jan 16, 2012 at 5:48 AM, Louis Carreiro carreir...@gmail.com wrote:
 Hey all!

 I'm not sure if this went out the first time I sent it so I apologize now if
 it's a duplicate.

 I've been banging my head against the wall for a while (almost 18 hours
 today alone) with this one... I migrated our incomming T1's from the Option
 11 to our Asterisk box this morning. We have 1 local T1 and 2 long distance
 T1's. The local T1 went over with out a hitch. The problem is with my 2 long
 distance T1's. The switch on the other end is a DMS250 I'm told so I set
 Asterisk to DMS100 and got the timing, framing, etc all set. Well, the D
 channels came up so thats good. I started getting dropped calls every once
 in a while. I did a debug on the spans and saw the following:

I have found that in most cases the easiest way to fix these issues is
to simply call the provider and ask them to switch it to NI2. Most of
them can do that while on the phone.



 PRI Span: 3
 PRI Span: 3  Protocol Discriminator: Q.931 (8)  len=40
 PRI Span: 3  TEI=0 Call Ref: len= 2 (reference 857/0x359) (Sent from
 originator)
 PRI Span: 3  Message Type: SETUP (5)
 PRI Span: 3  [04 03 80 90 a2]
 PRI Span: 3  Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info
 transfer capability: Speech (0)
 PRI Span: 3   Ext: 1  Trans mode/rate: 64kbps,
 circuit-mode (16)
 PRI Span: 3     User information layer 1: u-Law
 (34)
 PRI Span: 3  [18 04 e9 80 83 08]
 PRI Span: 3  Channel ID (len= 6) [ Ext: 1  IntID: Explicit  Other(PRI)
 Spare: 0  Exclusive  Dchan: 0
 PRI Span: 3    ChanSel: As indicated in following
 octets
 PRI Span: 3    Ext: 1  DS1 Identifier: 0
 PRI Span: 3    Ext: 1  Coding: 0  Number Specified
 Channel Type: 3
 PRI Span: 3    Ext: 0  Channel: 8 Type: CPE]
 PRI Span: 3  [20 02 00 e2]
 PRI Span: 3  Network-Specific Facilities (len= 2) [ Toll Free MEGACOM ]
 PRI Span: 3  [6c 0c 21 83 37 32 37 34 3033 34 30 37 34]
 PRI Span: 3  Calling Number (len=14) [ Ext: 0  TON: National Number (2)
 NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)

 The key part is the Ext: 1  DS1 Identifier: 0 part. That's when calls
 fail. Right now, all calls are coming in on span 3 and want to talk to
 Identifier 0 (span 2). If a call comes in on span 2 and requests Ext: 1
 DS1 Identifier: 1, it fails. I called Verizon and asked them what was going
 on. Turns out, its configured as a trunk group. The tech mentioned that I
 need to figure out how to set my identifiers on the group and I should be
 good to go. I've done a ton of research about chan_dahdi.conf and
 dahdi-channels.conf and I think the answer is trunk groups.

 I tried configuring a trunkgroup and set the primary dch to 24 and the bdch
 to 72 and then then spanmap'ed span 2 and 3 into group 1 (e.g. 2,1,0 and
 3,1,1) but I don't see anything when I do a dahdi show channels or a pri
 show spans or a pri show channels, not even the channels not in the
 group. If I delete the trunkgroup, all three commands return all the
 channels.

 I'm just curious if I'm going down the right path with trunkgroups for this
 or if there is something else to take care of the DS1 Identifier issue.

 So another quick look... when a sucessful call comes in it goes to DS1
 Identifier 0... the Asterisk CLI shows the following:

     -- Accepting call from '727403' to '890' on channel 0/11, span 2

 Is there a way to get the other span (span 3) to become channel 1/xx? So
 when a call comes in asking for DS1 Identifier 1 I see the following:

     -- Accepting call from '727403' to '890' on channel 1/12, span 3


 Thanks in advance everyone!

 Louis

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Re: [asterisk-users] Best non polycom SIP conference room phone

[C F]

On Sun, Jan 8, 2012 at 12:03 PM,  brya...@zktech.com wrote:
 Thank you for your responses. No where did I say I hate polycom phones. I 
 personally do not like their approach to sip as a company. Their audio 
 quality  is top notch but for me the rest leaves me wanting. Has anyone used 
 the newer snom conference room phone?


Oh. So its political?
They still make the best quality phones.

 Bryant Zimmerman

 On Jan 8, 2012, at 10:59 AM, C F shma...@gmail.com wrote:

 I find that the bottom line of all polycom haters is ones inability of
 comprehending the config files and not in its quality.
 However check out Panasonic. They make a sip conference phone.

 On 1/5/12, Carlos Alvarez car...@televolve.com wrote:
 On Thu, Jan 5, 2012 at 5:10 PM, C F shma...@gmail.com wrote:

 On Thu, Jan 5, 2012 at 12:19 PM, Bryant Zimmerman brya...@zktech.com
 wrote:
 I am looking for a really good SIP conference room phone for use with
 asterisk. I do not like Polycom at all.

 You have a really bad taste.


 There was an interesting flamewar one day in the Asterisk IRC channel over
 Polycom love/hate.  We fall into the hate category here, and hope to never
 have to deal with them.  If there was an SPA-series conference phone, we'd
 all rejoice.

 --
 Carlos Alvarez
 TelEvolve
 602-889-3003


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Re: [asterisk-users] Best non polycom SIP conference room phone

[C F]

On Sun, Jan 8, 2012 at 3:38 PM, Carlos Alvarez car...@televolve.com wrote:


 On Sun, Jan 8, 2012 at 9:02 AM, C F shma...@gmail.com wrote:

 I find that the bottom line of all polycom haters is ones inability of
 comprehending the config files and not in its quality.


 We have no problem with their config files.  They are no worse than anything
 else, including the SPA series phones that we greatly prefer over the
 Polycom.  The Polycom phones simply are more effort and more time-consuming
 than the SPA phones, and some others (though there are worse phones).  We
 hate working with them for a wide variety of reasons, but the config files
 are certainly not one of them.


Time consuming in what way? Do you mind elaborating why you hate them?


 However check out Panasonic. They make a sip conference phone.


 I didn't know about the conference phone from them.  We sell their wireless
 phones and found them extremely annoying to learn to configure, with lots of
 quirks and bugs, but once they are working they are good.  Once you get to
 know the oddities and have a suitable provisioning server set up, deploying
 more is no problem.  Troubleshooting is annoying because the documentation
 is poor and there are lots of quirks/bugs/unexpected features.

Its a nice although expensive phone, kx-nt700.
http://www2.panasonic.com/webapp/wcs/stores/servlet/BTSModelDetail?storeId=11201catalogId=13051catGroupId=141510itemId=330235modelNo=KX-NT700displayTab=O



 User acceptance on the Panasonic is very good.

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 TelEvolve
 602-889-3003



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Re: [asterisk-users] Is it valid to Dial(DAHDI/g0/12345wwwww88888888) on an ISDN trunk?

[C F]

Exactly which IE message are you trying to push manually? you
shouldn't have to do that, it should be done in the configs for you.

On Mon, Jan 9, 2012 at 1:45 PM, Johann Steinwendtner
steinwendt...@gmx.net wrote:
 On 2012-01-09 17:46, Alex Villací­s Lasso wrote:

 I am trying to collect information regarding a bug report for Elastix
 (http://bugs.elastix.org/view.php?id=1146). In this bug, an user has
 asterisk-1.8.7 and dahdi-2.4.1.2. He is trying to make an
 outbound call through an ISDN trunk, by placing
 Dial(DAHDI/g0/12345w) in order to send 12345, then wait a
 period, then send . I am still waiting for a response on what
 particular
 telephony card he uses, and the kind of ISDN setup (T1/E1/BRI) being used,
 but I want to know: Is this dialstring expected to work with an ISDN trunk?
 If so, are there any configurations that might
 cause it to stop working? The user claims that this same dialstring worked
 with Elastix 1.6 which had dahdi-2.2.0.2 and asterisk-1.4.26.1.

 Some additional information: the user reports that the dial attempt fails
 with hangup cause 28. From
 http://helpdesk.netcentral.co.uk/index.php?_m=knowledgebase_a=viewarticlekbarticleid=293

 http://helpdesk.netcentral.co.uk/index.php?_m=knowledgebase_a=viewarticlekbarticleid=293
 [^
 http://helpdesk.netcentral.co.uk/index.php?_m=knowledgebase_a=viewarticlekbarticleid=293]
 :

 Code No. 28 - invalid number format (address incomplete).
 This cause indicates that the called party cannot be reached because the
 called party number is not in a valid format or is not complete.

 Is it actually possible that the code is trying to send a string of 'w's
 through the ISDN link? Or am I misunderstanding?


 I'm using 'w' to force sending the 'sending complete' IE in an ISDN setup
 message.
 But I don't know the usage of multiple 'w' in the  dialstring.


 regards

 Hans


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Re: [asterisk-users] Connecting to an Old Phone System

[C F]

What type of phone system?
And what type of connectivity are you trying to give the old pbx?

On 1/6/12, Dan Journo d...@keshercommunications.com wrote:
 Hi,

 This is not strictly an asterisk questions, but... ive got a client with an
 old digital pbx phone systems connected to an isdn30e line.

 I've been shown a sip gateway that can connect to asterisk on one side, and
 also has an ISDN30e socket that the old phone system can connect to. But
 it's a bit pricey.

 Is there such a thing as an ISDN30e PCI card which can be used with a copy
 of Asterisk, that can act like a voip gateway between the old phone system,
 and our asterisk box?
 Or can anyone recommend a gateway that isn't too expensive?
 They use 8 channels of the isdn30e

 Many thanks
 Dan

 Dan Journo
 Kesher Communications (UK)
 Business Phone Systemshttp://www.keshercommunications.com/ | Hosted
 PBXhttp://www.keshercommunications.com/hostedpbx.html



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Re: [asterisk-users] Best non polycom SIP conference room phone

[C F]

I find that the bottom line of all polycom haters is ones inability of
comprehending the config files and not in its quality.
However check out Panasonic. They make a sip conference phone.

On 1/5/12, Carlos Alvarez car...@televolve.com wrote:
 On Thu, Jan 5, 2012 at 5:10 PM, C F shma...@gmail.com wrote:

 On Thu, Jan 5, 2012 at 12:19 PM, Bryant Zimmerman brya...@zktech.com
 wrote:
  I am looking for a really good SIP conference room phone for use with
  asterisk. I do not like Polycom at all.

 You have a really bad taste.


 There was an interesting flamewar one day in the Asterisk IRC channel over
 Polycom love/hate.  We fall into the hate category here, and hope to never
 have to deal with them.  If there was an SPA-series conference phone, we'd
 all rejoice.

 --
 Carlos Alvarez
 TelEvolve
 602-889-3003


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Re: [asterisk-users] Best non polycom SIP conference room phone

[C F]

On Thu, Jan 5, 2012 at 12:19 PM, Bryant Zimmerman brya...@zktech.com wrote:
 I am looking for a really good SIP conference room phone for use with
 asterisk. I do not like Polycom at all.

You have a really bad taste.

 What would you all recommend? I have
 to be able to get them in the US. I found several that looked good but could
 not get them. And yes cost does matter but quality is the most important
 thing.

Then go with Polycom.




 Thanks

 Bryant

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Re: [asterisk-users] A new hack?

[C F]

On Tue, Dec 6, 2011 at 5:19 AM, Hans Witvliet aster...@a-domani.nl wrote:
 On Mon, 2011-12-05 at 18:51 -0800, Steve Edwards wrote:
 snip

 Your security needs depends on your environment. At this point in time,
 all of the hosts I manage for my clients exist in very limited
 environments and have very small attack surfaces. They are racked in
 secure data centers. They only accept SIP from clients with static IP
 addresses that we have an existing business relationship with. They only
 accept SSH connections from me. They only accept HTTP connections from me
 and my boss. That's about it. I don't see where F2B adds much value for
 me.

 *) Lots of admins think they can't limit access to servers because they
 have 'mobile' users. Your users probably don't need to access your servers
 from every single place on the Internet. If your users don't come from
 China, North Korea, Iran, etc, you can block entire regions with a few
 rules and eliminate 80% of probes and attacks from reaching your servers
 in the first place. Apologies in advance if you happen to live in some of
 these regions -- feel free to `s/China, North Korea, Iran/United States,
 Canada, England/g`


 Perhaps an other suggestion.
 If they are true road warriors, i presume they are capable of setting
 up an vpn to the company.
 In that case, only allow  registrations/calls through the secured
 tunnel. Then it's not any concern to asterisk.

 And if they can breach your tunnel, you have something else to worry
 about.

Well, that means opening up VPN connections from everywhere. Thats why
I suggested turning off the server completely.

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Re: [asterisk-users] A new hack?

[C F]

On Fri, Dec 2, 2011 at 11:35 AM, Jim Lucas li...@cmsws.com wrote:
 On 11/26/2011 5:00 PM, C F wrote:
 On Sat, Nov 26, 2011 at 7:50 AM, Gordon Henderson
 gordon+aster...@drogon.net wrote:
 On Sat, 26 Nov 2011, Terry Brummell wrote:

 Install  Configure Fail2Ban then the host will be blocked from
 connecting.  And no, it's not new.

 I don't need Fail2Ban, thank you. But your advice might be useful to others.

 Why is that?
 Even if they don't compromise an account they are still using your
 bandwidth and resources on your machine.


 How is using Fail2Ban less resource intensive then me writing (by hand) 
 iptable
 rules?

Sorry I wasnt very clear in my first writing, I'll try to clarify.
Using iptables only detects one type of attack (aggressive
connections). While his machines might be secure enough to allow any
other attacks and still not compromise his machine, iptables will
still allow them thru and therefore the attack will be using his
bandwidth/resources, with f2b one can add as many rules as/when they
arrive.


 Also, since both methods involve the use of iptables, where exactly is the
 bandwidth savings?

In detection.


 --
 Jim Lucas

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Re: [asterisk-users] A new hack?

[C F]

On Mon, Dec 5, 2011 at 9:51 PM, Steve Edwards asterisk@sedwards.com wrote:
 (This horse just won't stay dead...)

 My apologies if I mis-attribute who wrote what.

 On Fri, Dec 2, 2011 at 11:35 AM, Jim Lucas li...@cmsws.com wrote:


 How is using Fail2Ban less resource intensive then me writing (by hand)
 iptable rules?


 On Mon, 5 Dec 2011, C F wrote:

 Sorry I wasnt very clear in my first writing, I'll try to clarify. Using
 iptables only detects one type of attack (aggressive connections). While his
 machines might be secure enough to allow any other attacks and still not
 compromise his machine, iptables will still allow them thru and therefore
 the attack will be using his bandwidth/resources, with f2b one can add as
 many rules as/when they arrive.


 I think you are over-generalizing.

 You can write iptables rules to detect and respond to many types of attacks.

Possible. But working off the logs makes lots more sense for creating
more accurate to the point rules, and to mention on the fly.


 Since F2B is just an automated front end to iptables you can have as many
 rules as you need with or without F2B. Also, since packets are 'stopped' at
 the same place (iptables) any bandwidth savings would only be to services
 that you are running that either aren't or can't* be nailed down.

You didn't get my point. If someone is trying to exploit some type of
dialplan hack in slow motion. iptables will probably not detect it and
your machine is secure enough that the exploit doesn't work, but the
script kiddie behind the attack doesn't know that and keeps trying.
Your wasting resources and bandwidth. With f2b you can have him added
to iptables after the first try. Once all packets are dropped from
that IP, while the attacker is still using resources/bandwidth while
trying after a while they will stop as all packets are dropped. The
reason they are trying is because it wasn't blocked but now that it is
they will stop.


 Also, since both methods involve the use of iptables, where exactly is
 the bandwidth savings?


 In detection.


 How about 'in responding to an attack your iptables rules don't already
 mitigate and you do have F2B rules defined for?' 'Detecting' an attack means
 close to nothing if you don't respond to it :)

I think you are just explaining my point. Correct me if I'm wrong.


 I'm not hating on F2B, it's just not a silver bullet nor is it appropriate
 for all environments.

Agreed, like another poster said, its the easy way out since it's an
easy front end. The only reason for this thread is because someone
mentioned he doesn't *need* it.


 Your security needs depends on your environment. At this point in time, all
 of the hosts I manage for my clients exist in very limited environments and
 have very small attack surfaces. They are racked in secure data centers.

Speaking of which, how secure? I have biometrics access to about a
dozen such centers. Once inside the center how hard is it really to do
what you want?

 They only accept SIP from clients with static IP addresses that we have an
 existing business relationship with. They only accept SSH connections from
 me. They only accept HTTP connections from me and my boss. That's about it.
 I don't see where F2B adds much value for me.

Well others keep their servers shut. While I'm sarcastic, I'm also
trying to say its way to overdone. A good IDS/IPS will do, there is
really no reason to this. Except in environments that require it, in
my opinion national infrastructure etc.


 *) Lots of admins think they can't limit access to servers because they have
 'mobile' users. Your users probably don't need to access your servers from
 every single place on the Internet. If your users don't come from China,
 North Korea, Iran, etc, you can block entire regions with a few rules and
 eliminate 80% of probes and attacks from reaching your servers in the first
 place. Apologies in advance if you happen to live in some of these regions
 -- feel free to `s/China, North Korea, Iran/United States, Canada,
 England/g`

 --
 Thanks in advance,
 -
 Steve Edwards       sedwa...@sedwards.com      Voice: +1-760-468-3867 PST
 Newline                                              Fax: +1-760-731-3000

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Re: [asterisk-users] A new hack?

[C F]

On Thu, Dec 1, 2011 at 8:15 AM, Gordon Henderson
gordon+aster...@drogon.net wrote:
 On Tue, 29 Nov 2011, C F wrote:

 BTW, you were just proven wrong, you need it for this hack.

 In addition to the few hundred protected asterisk installations I run, I
 also run a few honeypots.

Protected? You don't know that until the next hack comes out.

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Re: [asterisk-users] A new hack?

[C F]

On Sun, Nov 27, 2011 at 8:47 AM, Gordon Henderson
gordon+aster...@drogon.net wrote:
 On Sat, 26 Nov 2011, C F wrote:

 On Sat, Nov 26, 2011 at 7:50 AM, Gordon Henderson
 gordon+aster...@drogon.net wrote:

 On Sat, 26 Nov 2011, Terry Brummell wrote:

 Install  Configure Fail2Ban then the host will be blocked from
 connecting.  And no, it's not new.

 I don't need Fail2Ban, thank you. But your advice might be useful to
 others.

 Why is that?
 Even if they don't compromise an account they are still using your
 bandwidth and resources on your machine.

 Linux has excellent built-in subsystems to control firewalling and so on
 without resorting to external programs. It's called iptables. If you know
 how to use them, then using an external resource such as fail2ban is
 unneccessary.
So its not that you don't need it, but you use something else.
BTW, you were just proven wrong, you need it for this hack.

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Re: [asterisk-users] A new hack?

[C F]

On Mon, Nov 28, 2011 at 10:57 AM, Tom Browning ttbrown...@gmail.com wrote:
 On Sun, Nov 27, 2011 at 8:47 AM, Gordon Henderson
 gordon+aster...@drogon.net wrote:
 Linux has excellent built-in subsystems to control firewalling and so on
 without resorting to external programs. It's called iptables. If you know
 how to use them, then using an external resource such as fail2ban is
 unneccessary.

 That's like saying you don't need FreePBX because you have this thing
 called Asterisk.

Very well put.

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Re: [asterisk-users] A new hack?

[C F]

On Sat, Nov 26, 2011 at 7:50 AM, Gordon Henderson
gordon+aster...@drogon.net wrote:
 On Sat, 26 Nov 2011, Terry Brummell wrote:

 Install  Configure Fail2Ban then the host will be blocked from
 connecting.  And no, it's not new.

 I don't need Fail2Ban, thank you. But your advice might be useful to others.

Why is that?
Even if they don't compromise an account they are still using your
bandwidth and resources on your machine.

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Re: [asterisk-users] DID and how the caller id will appear

[C F]

On Tue, Oct 18, 2011 at 8:46 PM, bilal ghayyad bilmar...@yahoo.com wrote:
 Dear;

 By the way, the asterisk version that I have is 1.8.4.2 and DAHDI version is 
 2.4.1.2

 Here I would like to mention the following:

 1) As per the telecom provider, they said they openned for us all the digits 
 to send (two digits, 4 digits, all the digits ...) as they said.

With no screening? Make sure they allow you to set to at least the
DIDs that are allowed in.


 2) The caller id appear for me at the CLI as: 5631030 (and actually this is 
 the primary key).

 3) I am settings in the systems.conf file the loadzone to be uk us and the 
 defaultzone to be uk, does this effect? Maybe I have to set us.

 4) I am sending the numbers within my range it is from 5631030 to 5631059, 
 and actually I tried to set the caller id to be 40, 1040, 5621040, and 
 065621040

 5) It is E1 ISDN.

 6) About the:

 Set(CALLERID(pres)=allowed)

 How can I know the updated changes?
 By the way: is it important to be set?

 7) How about the Set(CALLERID(num)=1040), is it in the right syntax or also 
 need to be updated? Also, should I use Set(CALLERID(num)=1040) or 
 Set(CALLERID(num)=1040)?

 What other factors I am missing? What I have to check?

 Regards
 Bilal

 ---

 On Tuesday 18 October 2011, bilal ghayyad wrote:
  We contacted the Telecom provider and they confirmed
 multiple times that
  the DID service is enabled, but again still the caller
 id does not appear
  as we need (it is always appearing as the primary
 number). I tried to set
  the CALLERDID(num) to be 40, 1040, 5631040 and
 065631040 without any
  result.
 
  Where could be the problem?
 
  Maybe I have to use CALLERID(ANI)? Well, how it should
 be written exactly?
 
  What other things I have to do or I can try to resolve
 the problem?

 What format do your incoming caller IDs come
 in?   (use a NoOp() or Verbose()
 statement to write to the log.)  One would expect
 outgoing caller IDs to be
 set in the same format  (but speak to someone at the
 telco for confirmation of
 this).  Our provider  (in the UK)  expects
 our outgoing caller IDs to include
 the STD code, but without the initial 0.  So 065631040
 would be sent as
 65631040 .


 Are you definitely allowed to set your own caller ID?
 It may well be
 restricted to one of a range of presentation numbers
 assigned to you.


 One other thing:  CALLERPRES() has been deprecated for
 awhile now, so in newer
 Asterisk versions you may need to write
 Set(CALLERID(pres)=allowed) instead of
 Set(CALLERPRES()=allowed .

 --
 AJS

 Answers come *after* questions.




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Re: [asterisk-users] Maybe slightly OT but..

[C F]

On Mon, Oct 10, 2011 at 10:31 PM, linux guy linuxguy...@gmail.com wrote:
 On Mon, Oct 10, 2011 at 8:08 PM, Andres and...@telesip.net wrote:

 I would recommend Acrobits.  Not free but only a few bucks.  It works fine
 with ATT 3G.

 This begs the question... which is more expensive (and where)...
 making a regular cell call or making a SIP call over 3G ?

Really depends if you are talking quality or ?
Anyhow, I doubt anyone is getting fine with ATT 3G try holding the
call for more than 2-3 minutes, they have a terrible 3G network. I use
them and the only reason I haven't yet dumped them is because of the
ability of browsing and talking at the same time. But with the amount
of garbage I have to put up with them, I doubt that one benefit is
going to prevail.






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Re: [asterisk-users] Call does not pass through

[C F]

Can you please post the relevant parts of extensions.conf? As well as
a CLI output of when you dial and it fails?

On 9/26/11, Malvin Rito mr...@mail.altcladding.com.ph wrote:
 Hi list,
 My call does not pass through on the first dial, I have to redial again the
 number for the call to  pass through. I'm not sure if the problem is on my
 voip proovider or my asterisk server setup. Any thoughts pls?

 Regards,
 Malvin


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Re: [asterisk-users] DID and the Caller ID for outgoing

[C F]

Did you add the Set(CALLERID(num) as I have previously pointed out?

On 9/25/11, bilal ghayyad bilmar...@yahoo.com wrote:
 Hi All;

 I do not know if the SEP_Mac_address.cnf.xml of the cisco file is also has
 effecting on the DID and Caller ID to appear at the destination, because I
 found the following:


  localCfwdEnabletrue/localCfwdEnable
  semiAttendedTransfertrue/semiAttendedTransfer
  anonymousCallBlock2/anonymousCallBlock
  callerIdBlocking2/callerIdBlocking

 So, does the callerIdBlocking2/callerIdBlocking is effecting on
 displaying the caller id at the destination?

 What does it mean the value to be 2?

 Because I am placing the callerid=5631040 (and I tried callerid=5631040
 also) in the sip.conf for the Cisco IP Phone, and no success, it is always
 displaying the primary number which is: 5631030

 So, I started think if the callerIdBlocking2/callerIdBlocking at this
 setting file is effecting? Or if there is any parameter is effecting?

 Any help?

 Regards
 Bilal



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Re: [asterisk-users] DID and how the caller id will appear

[C F]

Confirm with your provider that allow you to set caller id on outbound.


On Sun, Sep 25, 2011 at 1:59 PM, bilal ghayyad bilmar...@yahoo.com wrote:
 Dear;

 By the way, the asterisk version is: 1.8.4.2

 Yes I tried Set(CALLERID(num)=5631040) as shown in the below dialing, and no 
 success. Also I tried Set(CALLERID(num)=1040) and I tried 
 Set(CALLERID(num)=065631040) as the city code is 06 and when we call any 
 mobile, it is appearing 065631040, but all of this did not work.

 Do I have to use SetCallerPres? What is the value?

 The E1 is located in Jordan and it is PRI with 30 channels. Is there any 
 thing need to be set other than Set(CALLERID(num)? I am afraid that I have to 
 set a specific value for SetCallerPress !

 Well, I have also a question: What should I set the callerid when I am 
 configuring the IP Phone in the sip.conf?

 By the way: what is the difference between using Set(CALLERID(num)=5631040) 
 and the callerid in the sip.conf?

 Kindly find below my dialing plan:

 exten = _90Z,1,Set(CALLERID(num)=5631040)
 exten = 
 _90Z,2,MixMonitor(${CHANNEL}${EXTEN:1}${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)}.wav)
 exten = _90Z,3,Dial(${PSTNTRUNK}/${EXTEN:1})
 exten = _90Z,4,Playback(vm-nobodyavail)
 exten = _90Z,5,Hangup()
 exten = _90Z,104,Congestion() ; if no channel available
 exten = _90Z,105,Hangup()


 ---



 Set(CALLERID(num)=5631040)
 add this before the Dial command.

 On Sat, Sep 24, 2011 at 4:03 PM, bilal ghayyad bilmar...@yahoo.com
 wrote:
  Hi All;
 
  The DID range that we took from the telecom starts
 from 1030 and end by 1059, now whenever we place a call, the
 destination see the number 5631030. I gave the phone
 extension 1040, and when I call, still the destination see
 the number is 5631030?
 
  Kindly find below the configuration of the extension
 1040, please what I have also to configure so when this
 extension make a call, the destination see it 5631040?
 
  [1040]
  type=friend
  host= dynamic
  callerid=1040
  disallow=all
  allow=alaw
  allow=ulaw
  allow=g729
  context=External
  dtmfmode=auto
  nat=no
  qualify=no
  canreinvite=yes
  username=1040
  secret=***
 
  Regards
  Bilal


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Re: [asterisk-users] DID and how the caller id will appear

[C F]

Set(CALLERID(num)=5631040)
add this before the Dial command.

On Sat, Sep 24, 2011 at 4:03 PM, bilal ghayyad bilmar...@yahoo.com wrote:
 Hi All;

 The DID range that we took from the telecom starts from 1030 and end by 1059, 
 now whenever we place a call, the destination see the number 5631030. I gave 
 the phone extension 1040, and when I call, still the destination see the 
 number is 5631030?

 Kindly find below the configuration of the extension 1040, please what I have 
 also to configure so when this extension make a call, the destination see it 
 5631040?

 [1040]
 type=friend
 host= dynamic
 callerid=1040
 disallow=all
 allow=alaw
 allow=ulaw
 allow=g729
 context=External
 dtmfmode=auto
 nat=no
 qualify=no
 canreinvite=yes
 username=1040
 secret=***

 Regards
 Bilal

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Re: [asterisk-users] Problem with multiple sip-peers against the same host

[C F]

Can you please post:
1. Relevant sip.conf
2. sip debug when trying to make a call?

On Thu, Sep 22, 2011 at 7:26 PM, David Björkevik da...@dynamore.se wrote:
 Dear list,

 We are switching to a new provider for SIP-trunks. We have 20 numbers,
 each defined as a separate SIP peer.

 With the old provider everything works.

 When switching to the new provider's account data, it only works as long
 as I only define one of the accounts.  If multiple accounts are defined,
 I can only place outgoing calls on one of them, for the other(s)
 authentication fails, FORBIDDEN.

 It is almost like Asterisk is using just one of the defined passwords to
 authenticate all peers on that host.

 Any input is very appreciated.

 Regards
 David Björkevik, Engineer


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Re: [asterisk-users] Ghost DID in System

[C F]

Without your dialplan there isnt much that can be done to help.
Can you please post your relevant dialplans?
Whats voip1 and voip2?
When you say outside the voip system call goes thru, to where?
Who has the number currently?
Any sip debug you care sharing?


On Mon, Sep 19, 2011 at 6:51 PM, Aaron Krohn akr...@ewebforce.net wrote:
 This is going to sound ridiculous, but there appears to be a ghost DID in
 our system. We are going to get the number ported to us, but it has not
 happened yet. From a phone outside of our voip system, the call still goes
 through. When calling the did from a phone within our system, there is just
 dead air.

 In the asterisk CLI, I can see our primary server, voip1 trying to do pass
 the call to voip2 after it complains about not knowing what to do with the
 call.

 I have removed all references to this number from all dialplans and sip-did
 lists and restarted many times. I simply don't understand why our voip1
 server believes it should try to route the call instead of passing it to the
 outside world. Does anyone have an explanation or know where I could look?
 (dialplans, obviously =)


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Re: [asterisk-users] how to add-edit-delete entery into asterisk conf files

[C F]

Why php? Isn't vi the only way?

On Fri, Sep 2, 2011 at 7:28 AM, virendra bhati virbh...@gmail.com wrote:
 Hi list,

 I want ot do basic work (add-edit-delete) into asterisk configuration files,
 like sip.conf, manager.conf,musiconhold.conf etc.

 Please guide me how to configure all these files from from AMI connection. I
 am able to login into AMI from Login action but I want to do more task in to
 it.

 AMI login:-

 login.php

 ?php
 $socket = fsockopen(127.0.0.1,5038, $errno, $errstr, 30);
 fputs($socket, Action: Login\r\n);
 fputs($socket, UserName: root\r\n);
 fputs($socket, Secret: energy\r\n\r\n);
 ?
 AMI command:-

 Below commands are for musiconhold.conf. I want to add new MOH context into
 it.
 ?php
 include(login.php);
   fputs($socket, Action: UpdateConfig\r\n);
   fputs($socket, Filename: musiconhold.conf\r\n);
   fputs($socket, Srcfilename: musiconhold.conf\r\n);
   fputs($socket, Dstfilename: musiconhold.conf\r\n);
   fputs($socket, Action-00: newcat\r\n);
   fputs($socket, Cat-00: bhavik\r\n);
   fputs($socket, mode: files\r\n);
   fputs($socket, directory: /var/lib/asterisk/moh\r\n);
   fputs($socket, Reload: yes\r\n);
   fputs($socket, ActionID: 9873497149817\r\n);
   fputs($socket, Action: Logoff\r\n\r\n);

 ?

 After doing all no success :((


 -
 Thanks and regards

  Virendra Bhati
 +91-9172341457
 Software Engineer


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Re: [asterisk-users] Faxes suddenly failing

[C F]

I think you should change the subject line to:
Faxes suddenly worked for 2 weeks.

On Wed, Aug 31, 2011 at 3:49 PM, Tim King tim.compnetw...@gmail.com wrote:
 I realize that faxing is not great with voip but here is my confusion. I
 have been working on a web based fax system for 2 weeks. During this time I
 have sent over 100 2 page faxes without any errors. Now today as things are
 finally completed I can not seem to get any fax to go through unless it is a
 1 page cover only. Anyone able to tell the issue from this debug output?

    -- Channel 'SIP/MyVoipProvider-0046' FAX session '12' started
     -- FAX handle 0: [ 000.38 ], STAT_EVT_STRT_RX   st: IDLE
 rt: IDLENSRX
     -- FAX handle 0: [ 000.000184 ], STAT_EVT_RX_HW_RDY st: WT_RX_HW_RDY
 rt: RRDYNHRY
     -- FAX handle 0: [ 000.000504 ], P30EVN_RECEIVE_STARTED
     -- FAX handle 0: [ 000.000538 ], STAT_INFO_CSI
     -- FAX handle 0: [ 000.000568 ], STAT_INFO_DIS
     Channel 'SIP/MyVoipProvider-0046' fax session '12', [
 000.091837 ], stack sent 5 frames (100 ms) of energy.
     Channel 'SIP/MyVoipProvider-0046' fax session '12', [
 000.160248 ], stack sent 3 frames (60 ms) of silence.
     Channel 'SIP/MyVoipProvider-0046' fax session '12', [
 000.960201 ], channel sent 48 frames (960 ms) of silence.
     Channel 'SIP/MyVoipProvider-0046' fax session '12', [
 000.979464 ], channel sent 1 frames (20 ms) of energy.
     Channel 'SIP/MyVoipProvider-0046' fax session '12', [
 003.157848 ], stack sent 150 frames (3000 ms) of energy.
     Channel 'SIP/MyVoipProvider-0046' fax session '12', [
 003.219814 ], stack sent 3 frames (60 ms) of silence.
     -- FAX handle 0: [ 005.240927 ], STAT_EVT_TX_V21_DONE   st: WT_DIS_RSP
 rt: WDSRNT21
     Channel 'SIP/MyVoipProvider-0046' fax session '12', [
 005.579811 ], stack sent 118 frames (2360 ms) of energy.
     Channel 'SIP/MyVoipProvider-0046' fax session '12', [
 006.481179 ], channel sent 275 frames (5500 ms) of silence.
     Channel 'SIP/MyVoipProvider-0046' fax session '12', [
 007.801045 ], channel sent 66 frames (1320 ms) of energy.
     -- FAX handle 0: [ 007.800554 ], STAT_FRM_CRP
     -- FAX handle 0: [ 007.800586 ], STAT_EVT_CRP   st: WT_DIS_RSP
 rt: NT4X
     -- FAX handle 0: [ 007.800602 ], STAT_EVT_FSC_ERR   st: WT_DIS_RSP
 rt: UNEXPECT
     -- FAX handle 0: [ 011.012832 ], STAT_EVT_RX_TRN_END    st: WT_DIS_RSP
 rt: RXXXNFRX
     -- FAX handle 0: [ 011.012878 ], STAT_INFO_CSI
     -- FAX handle 0: [ 011.012905 ], STAT_INFO_DIS
     Channel 'SIP/MyVoipProvider-0046' fax session '12', [
 011.152812 ], stack sent 279 frames (5580 ms) of silence.
     -- FAX handle 0: [ 013.179561 ], STAT_EVT_TX_V21_DONE   st: WT_DIS_RSP
 rt: WDSRNT21
     Channel 'SIP/MyVoipProvider-0046' fax session '12', [
 013.471827 ], stack sent 116 frames (2320 ms) of energy.
     Channel 'SIP/MyVoipProvider-0046' fax session '12', [
 014.260642 ], channel sent 323 frames (6460 ms) of silence.
     -- FAX handle 0: [ 016.119786 ], STAT_INFO_TSI
     Channel 'SIP/MyVoipProvider-0046' fax session '12', [
 016.460661 ], channel sent 110 frames (2200 ms) of energy.
     -- FAX handle 0: [ 016.460394 ], STAT_INFO_DCS
     -- FAX handle 0: [ 016.460431 ], STAT_EVT_DCS   st: WT_DIS_RSP
 rt: WDSRNDCS
     -- FAX handle 0: [ 016.460449 ], STAT_NEG_V17_14400
     -- FAX handle 0: [ 016.460464 ], STAT_NEG_MH
     -- FAX handle 0: [ 016.460476 ], STAT_NEG_A4
     -- FAX handle 0: [ 016.460488 ], STAT_NEG_RES_204x196
     -- FAX handle 0: [ 016.460500 ], STAT_NEG_ECM
     -- FAX handle 0: [ 016.460514 ], STAT_EVT_SW_ECM    st: WT_DIS_RSP
 rt: WDSRNSWE
     Channel 'SIP/MyVoipProvider-0046' fax session '12', [
 016.540315 ], channel sent 4 frames (80 ms) of silence.
     -- FAX handle 0: [ 016.800906 ], STAT_EVT_RX_IMG_STRT   st: RCV_ECM_TRN
 rt: UNEXPECT
     Channel 'SIP/MyVoipProvider-0046' fax session '12', [
 019.700543 ], channel sent 158 frames (3160 ms) of energy.
     -- FAX handle 0: [ 019.759984 ], STAT_EVT_RX_TRN_END    st: RCV_ECM_TRN
 rt: RTCFNERT
     -- FAX handle 0: [ 019.760071 ], STAT_FRM_CFR
     Channel 'SIP/MyVoipProvider-0046' fax session '12', [
 019.912812 ], stack sent 322 frames (6440 ms) of silence.
     -- FAX handle 0: [ 020.957834 ], STAT_EVT_TX_V21_DONE   st: RCV_ECM_STRT
 rt: RECMNT21
     Channel 'SIP/MyVoipProvider-0046' fax session '12', [
 021.278809 ], stack sent 68 frames (1360 ms) of energy.
     Channel 'SIP/MyVoipProvider-0046' fax session '12', [
 022.261160 ], channel sent 128 frames (2560 ms) of silence.
     -- FAX handle 0: [ 022.517880 ], STAT_EVT_RX_IMG_STRT   st: RCV_ECM_STRT
 rt: RECMNSRI
     -- FAX handle 0: [ 022.517982 ], P30EVN_PHASE_C
     -- FAX handle 0: [ 022.517998 ], P30EVN_DOC_START
     -- FAX handle 0: [ 022.518429 ], P30EVN_PAGE_START
     Channel 'SIP/MyVoipProvider-0046' fax 

Re: [asterisk-users] Dragging the dialup customers along, possible?

[C F]

On Wed, Aug 31, 2011 at 9:25 AM, Kevin P. Fleming kpflem...@digium.com wrote:
 On 08/29/2011 10:32 PM, C F wrote:

 On Mon, Aug 29, 2011 at 4:32 PM, Cary Fitchca...@usawide.net  wrote:


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
 Sent: Monday, August 29, 2011 3:18 PM
 To: t...@ewebforce.net; Asterisk Users Mailing List - Non-Commercial
 Discussion
 Subject: Re: [asterisk-users] Dragging the dialup customers along,
 possible?

  From what you are asking it appears that you are trying to run similar
 to a fax (modulation and demodulation) over VoIP.
 Try again, the fact that you succeeded twice was pure luck, and as far
 as I understand that didn't even work out.
 Switch back to TDM. Your dial up modems want that magic thing called
 timing and no jitter that only TDM will give you.

 ===


 So you want to develop the equivalent of T.38 for dial up?

 It already exists; it's called V.150.

Wow, thanks for the info.


 --
 Kevin P. Fleming
 Digium, Inc. | Director of Software Technologies
 Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] Avaya to Asterisk Voice mail

[C F]

What exactly is your setup?

On Tue, Aug 30, 2011 at 10:44 AM, Dustin fails wff...@gmail.com wrote:
 Has anyone have Avaya setup to ring to Asterisk voice mail over an analogue
 line. The issue I am having is Avaya is sending the originating caller id
 not the station id so Asterisk see that originating id so I can't route the
 call correctly in Asterisk.

 Thanks!

 Dustin

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Re: [asterisk-users] Linksys/Cisco 504G randomly restarts

[C F]

On Tue, Aug 16, 2011 at 10:27 AM, Jose Flores Galicia floj...@gmail.com wrote:
 I have seen this with last firmware.

 You need to change those 2 parameters to get a non-autoreboot scenario:

   Resync_At__HHmm_ ua=na0001/Resync_At__HHmm_
   Resync_Periodic ua=na/Resync_Periodic

Thanks, looks like this did the trick.



 Whenever it resync profile, it reboots, so setting resync to an hour nobody
 uses it and unseting resync periodicaly solve it for me.

 Jose Flores Galicia
 BriefCode  Code Based Training


 2011/8/15 C F shma...@gmail.com

 On Mon, Aug 15, 2011 at 2:54 PM, Vahan Yerkanian va...@arminco.com
 wrote:
  On 8/15/11 10:46 PM, C F wrote:
 
  I have 3 Linksys/Cisco 504G phones they keep restarting at what seems
  to be random. Sometimes as short as 6 minutes.
  FW version is 7.4.3a
 
  I have searched and tried disabling FW check and all related settings.
  I also extended all the default 3600 resync checks to a lot longer.
 
  TIA
  CF
 
  Hi,
 
  Try upgrading to the latest version. I have tens of 504G operating
  without
  any problems.

 That is going to be my next step. Thank you

 
  How are you powering these phones? I had a case when a PoE switch was
  experiencing short-circuit problems on a badly wired cable, and was
  unable
  to provide enough current to power the phones on the other ports.
  Replacing
  the faulty cable fixed the problem. You can always try to power the
  phones
  using 5volts DC, 2A center pin positive power source and see if the
  problem
  persists.

 They are all locally powered. I can see in asterisk CLI that its not a
 power issue since they are unregistering before shutdown.

 Thanks again.

 
  Also I have a Linksys SPA-941 that has a public IP and reboots itself
  whenever someone tries to bruteforce into it by sending tons of sip
  registers :)
 
  HTH,
  Vahan
 
 
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Re: [asterisk-users] Dragging the dialup customers along, possible?

[C F]

From what you are asking it appears that you are trying to run similar
to a fax (modulation and demodulation) over VoIP.
Try again, the fact that you succeeded twice was pure luck, and as far
as I understand that didn't even work out.
Switch back to TDM. Your dial up modems want that magic thing called
timing and no jitter that only TDM will give you.


On Mon, Aug 29, 2011 at 2:56 PM, Aaron Krohn akr...@ewebforce.net wrote:
 We have an Ascend Max router that has a PRI plugged into it, providing our
 current dialup users with web access. The PRI is no longer cost effective,
 so I've been tasked with converting it to something cheaper.

 We added a DID to our (existing) asterisk system (we have a couple dozen
 voip customers). We added an Adtran 908 to convert the VoIP signal into a
 virtual PRI for our MAX router to handle dialup calls.

 When dialing into the number with a modem, MAX sees the call, picks up and
 apparently tries to negotiate it, but eventually disconnects. It HAS,
 however, twice, successfully connected the call for a short time, but no
 browsing was possible. I've done some debugging output on the Adtran which
 seems to indicate that an RTP BYE command is received: TM.T01 01
 SipTM_Connected      rcvd SIP call-leg request: BYE ... This is the first
 difference between a debug output where the call connected and one that does
 not work. This is the one that doesn't work.

 TL;DR - Is it possible to do dial-up through Asterisk? Or is it like t38
 faxing that is 'possible' to get working, but not if you have other job
 responsibilities?

 Thanks!

 --
 Aaron Krohn
 Web Force Systems

 Business Office:
 131 Dillmont Drive, Suite 201
 Columbus, OH 43235
 Direct:  614-384-0019    Fax:  614-785-0871
 Tech Support / Help Desk Direct:  614-384-0020


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Re: [asterisk-users] Dragging the dialup customers along, possible?

[C F]

On Mon, Aug 29, 2011 at 4:32 PM, Cary Fitch ca...@usawide.net wrote:


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
 Sent: Monday, August 29, 2011 3:18 PM
 To: t...@ewebforce.net; Asterisk Users Mailing List - Non-Commercial
 Discussion
 Subject: Re: [asterisk-users] Dragging the dialup customers along, possible?

 From what you are asking it appears that you are trying to run similar
 to a fax (modulation and demodulation) over VoIP.
 Try again, the fact that you succeeded twice was pure luck, and as far
 as I understand that didn't even work out.
 Switch back to TDM. Your dial up modems want that magic thing called
 timing and no jitter that only TDM will give you.

 ===

 This is more of a whimsical statement than a scientific one, but I would
 think in today's world, there would be a real small box that would take in
 IP and put out TDM with good timing with a moderate buffering window.
 Obviously, the IP has to actually get to the box in a timely fashion, like
 today , but a TDM circuit has to be up also.

 A box that would take in IP data..., look for valid ascii, and otherwise
 put out TDM modem tones with no data content for 1 second and then pick up
 the data as it catches up.

So you want to develop the equivalent of T.38 for dial up?


 Better a laggy modem connection than no data at all.

 CF


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Re: [asterisk-users] Looking for ideas for nice home phone system

[C F]

The 824 is NOT discontinued.

On 8/23/11, John Novack jnov...@stromberg-carlson.org wrote:


 C F wrote:
 On Tue, Aug 23, 2011 at 5:21 PM, John Novack
 jnov...@stromberg-carlson.org  wrote:
 snip

 What do you mean by MD?


 MD is a common telephony term for Manufacture Discontinued

 John Novack

 --

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Re: [asterisk-users] Looking for ideas for nice home phone system

[C F]

Everything you mention for the NEC
system is available with the 824
but the Lan/Wan programming. In fact on
the 824 every port (to a
system max of 24) or analog as well as
proprietary. Support upto 4
doors with a chime and external relay
(switch cameras etc).


On 8/22/11, John Novack jnov...@stromberg-carlson.org wrote:
 NEC-DX-40 is another best buy
 Single pair phones
 2 analog station ports
 door box ports
 AND remote programming via LAN or WAN
 Voice mail available with or without email notification
 SIP gateway option

 Far superior to the TA-824 and Partners


 John Novack


 C F wrote:
 Panasonic KX-TA824
 Or the Panasonic KX-TAW848
 Or the Avaya Partner ACS 8.0


 On Mon, Aug 22, 2011 at 4:11 PM, Linuxguy123linuxguy...@gmail.com
 wrote:
 I'm looking for ideas for building a innovative, powerful home phone
 system.

 Something that does voicemail well, integrates cell phones into the
 house system, etc.

 I know there are a lot of details that need to be discussed, but lets
 leave it at that for now.

 What is everyone doing ?

 Thanks !



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Re: [asterisk-users] Looking for ideas for nice home phone system

[C F]

On Tue, Aug 23, 2011 at 5:21 PM, John Novack
jnov...@stromberg-carlson.org wrote:


 C F wrote:

 Everything you mention for the NEC
 system is available with the 824
 but the Lan/Wan programming. In fact on
 the 824 every port (to a
 system max of 24) or analog as well as
 proprietary. Support upto 4
 doors with a chime and external relay
 (switch cameras etc).

 Not quite - In addition to remote programming, the DSX series has single
 pair phones, and built in VM on a CF card. The 824 is an additional box.

The 824 also supports single pair phones on EVERY jack. As well as
their proprietary phones.
The 824 supports 3 types of Voicemail. 1. Provider based voicemail. It
will detect both FSK and stutter dial tone as a message indication and
display message waiting on the inside phones. 2. Built in voicemail
(called SVM). 3. external panasonic based voicemail (TVA50).

 Also the DSX has a SIP gateway option that in addition to supporting remote
 SIP phones, with latest software supports SIP trunks

Thats +1 for the DSX.

 The 824 also requires additional card(s) for CLID

Not out of the box, only if you add additional cards it needs a CLID
card as well for those additional lines.

 The analog/prop ports are a nice feature, and also allows both a prop
 Panasonic and POTS phone on a single extension
 I believe the 824 is also now MD, but some decent buys can be found in the

What do you mean by MD?

 used market.

 Any of the mentioned products would be a better solution for a newbee. Hang
 on the wall, do some quick programming, and forget it for the next 20 years,
 barring a power or lightning hit.
 No steep learning curve required. Certainly less expensive hardware

Exactly my point.


 John Novack

 On 8/22/11, John Novackjnov...@stromberg-carlson.org  wrote:

 NEC-DX-40 is another best buy
 Single pair phones
 2 analog station ports
 door box ports
 AND remote programming via LAN or WAN
 Voice mail available with or without email notification
 SIP gateway option

 Far superior to the TA-824 and Partners


 John Novack


 C F wrote:

 Panasonic KX-TA824
 Or the Panasonic KX-TAW848
 Or the Avaya Partner ACS 8.0


 On Mon, Aug 22, 2011 at 4:11 PM, Linuxguy123linuxguy...@gmail.com
 wrote:

 I'm looking for ideas for building a innovative, powerful home phone
 system.

 Something that does voicemail well, integrates cell phones into the
 house system, etc.

 I know there are a lot of details that need to be discussed, but lets
 leave it at that for now.

 What is everyone doing ?

 Thanks !



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Re: [asterisk-users] Looking for ideas for nice home phone system

[C F]

On Tue, Aug 23, 2011 at 6:59 PM, Linuxguy123 linuxguy...@gmail.com wrote:
 On Mon, 2011-08-22 at 22:13 -0400, C F wrote:
 Panasonic KX-TA824
 Or the Panasonic KX-TAW848
 Or the Avaya Partner ACS 8.0

 Are these Asterisk/VOIP based solutions ?

I tried using google translate to see if it will detect another
language here. Unless google thinks like humans I think it was written
in English.



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Re: [asterisk-users] Looking for ideas for nice home phone system

[C F]

Panasonic KX-TA824
Or the Panasonic KX-TAW848
Or the Avaya Partner ACS 8.0


On Mon, Aug 22, 2011 at 4:11 PM, Linuxguy123 linuxguy...@gmail.com wrote:
 I'm looking for ideas for building a innovative, powerful home phone
 system.

 Something that does voicemail well, integrates cell phones into the
 house system, etc.

 I know there are a lot of details that need to be discussed, but lets
 leave it at that for now.

 What is everyone doing ?

 Thanks !



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[asterisk-users] Linksys/Cisco 504G randomly restarts

[C F]

I have 3 Linksys/Cisco 504G phones they keep restarting at what seems
to be random. Sometimes as short as 6 minutes.
FW version is 7.4.3a

I have searched and tried disabling FW check and all related settings.
I also extended all the default 3600 resync checks to a lot longer.

TIA
CF

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Re: [asterisk-users] Linksys/Cisco 504G randomly restarts

[C F]

On Mon, Aug 15, 2011 at 2:54 PM, Vahan Yerkanian va...@arminco.com wrote:
 On 8/15/11 10:46 PM, C F wrote:

 I have 3 Linksys/Cisco 504G phones they keep restarting at what seems
 to be random. Sometimes as short as 6 minutes.
 FW version is 7.4.3a

 I have searched and tried disabling FW check and all related settings.
 I also extended all the default 3600 resync checks to a lot longer.

 TIA
 CF

 Hi,

 Try upgrading to the latest version. I have tens of 504G operating without
 any problems.

That is going to be my next step. Thank you


 How are you powering these phones? I had a case when a PoE switch was
 experiencing short-circuit problems on a badly wired cable, and was unable
 to provide enough current to power the phones on the other ports. Replacing
 the faulty cable fixed the problem. You can always try to power the phones
 using 5volts DC, 2A center pin positive power source and see if the problem
 persists.

They are all locally powered. I can see in asterisk CLI that its not a
power issue since they are unregistering before shutdown.

Thanks again.


 Also I have a Linksys SPA-941 that has a public IP and reboots itself
 whenever someone tries to bruteforce into it by sending tons of sip
 registers :)

 HTH,
 Vahan


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Re: [asterisk-users] FAX Issues

[C F]

On Tue, Aug 9, 2011 at 3:38 PM, Sassy Natan sas...@gmail.com wrote:
 Hi,
 I would like to make sure I got it right:
 1. Asterisk 1.4 doesn't support FAX support. It do however works if u sent
 fax from the PSTN and have anther FAX machine answer to it even if it is
 behind asterisk. This works like any regular phone, and as far as I know
 this mode known as T.38 pass through.

No its not T.38 pass through. T.38 pass through is only if the 2 end
points negotiated T.38, i.e. a provider that supports T.38 and an ATA
that supports T.38. What you are describing will most likely fail if
there is any VoIP in between, and should succeed over pure TDM using
digium or similar cards.

 2. If u want asterisk 1.4 to able to sent and receive emails you will have
 to patch the source code using the spandsp patches. There are some other
 ways to make this work like using  IAX, hylafax  but I need to know if this
 is true. This mode is what know to be as NVFAX detected

asterisk doesn't have the ability to send or receive emails. It does
have the ability to use something like sendmail to send/receive
emails.
I'm assuming its a typo and you meant faxes.
What exactly are you trying to accomplish? If you need to know
something go ahead and try/test it and report back. After all that is
what we all did to be able to answer it. Apparently the fact that
others tried it and told you that it works still warrants a
question:I need to know if this is true


 3. Now version 1.6 support Fax in a better way then 1.4. There is app_fax.c
 in the source code. Can someone please tell me what does this apps do?

I assume you meant than not then, but I'm not sure if its a typo, you
might just not know.
app_fax.c does faxing. The question is what are YOU trying to do?


 4. Version 1.8 as two modes: SpanDSP mode and some other method, what is
 the different between then can someone please help?

Whats the other mode?



 Thanks
 Sassy
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Re: [asterisk-users] FAX Issues

[C F]

On Tue, Aug 9, 2011 at 6:00 PM, Steve Totaro
stot...@totarotechnologies.com wrote:
 On Tue, Aug 9, 2011 at 5:21 PM, C F shma...@gmail.com wrote:
 On Tue, Aug 9, 2011 at 3:38 PM, Sassy Natan sas...@gmail.com wrote:
 Hi,
 I would like to make sure I got it right:
 1. Asterisk 1.4 doesn't support FAX support. It do however works if u sent
 fax from the PSTN and have anther FAX machine answer to it even if it is
 behind asterisk. This works like any regular phone, and as far as I know
 this mode known as T.38 pass through.

 No its not T.38 pass through. T.38 pass through is only if the 2 end
 points negotiated T.38, i.e. a provider that supports T.38 and an ATA
 that supports T.38. What you are describing will most likely fail if
 there is any VoIP in between, and should succeed over pure TDM using
 digium or similar cards.

 Fax was never officially supported by Asterisk/Digium using TDM cards.

 I works for the most part, but there is no comparison to a pure POTS
 line.  Much depends on the line quality, the sending and receiving fax
 devices.  Some places with good fax machines, PRIs coming in and some
 FXS ports would work just fine.

 Others, are a nightmare and make you look bad.

 Depending on your customer's fax needs, sometimes it is better to tell
 them to keep or have some POTS lines provisioned for reliable faxing
 and to back that up, bring up being able to reach 911 if the phone
 system is down and the company's liability, even though everyone has
 cell phones.  This is the best for TDM faxing with nothing fancy.

 I have sent people to www.trustfax.com

 TDM faxing is something that burned me more than once, so I don't suggest it.


 2. If u want asterisk 1.4 to able to sent and receive emails you will have
 to patch the source code using the spandsp patches. There are some other
 ways to make this work like using  IAX, hylafax  but I need to know if this
 is true. This mode is what know to be as NVFAX detected

 asterisk doesn't have the ability to send or receive emails. It does
 have the ability to use something like sendmail to send/receive
 emails.
 I'm assuming its a typo and you meant faxes.
 What exactly are you trying to accomplish? If you need to know
 something go ahead and try/test it and report back. After all that is
 what we all did to be able to answer it. Apparently the fact that
 others tried it and told you that it works still warrants a
 question:I need to know if this is true


 I have been advised that everyone must post their solutions to the
 list so that you don't have to test.  I think it was a self appointed
 list moderator.  People are required to come back, aknowledge that
 something worked and give credit.  Flaw: Just because something worked
 for someone doesn't mean that your solution wouldn't work better.

 Anyways, I am with you.  I learned this stuff with very little
 documentation except .conf examples and code.  Integration is where
 things get really fun and creative.  Logically testing and failing and
 testing until you find your solution is extremely rewarding (at least
 to me).  Making a Definity G3 and Asterisk fully integrate can cause
 severe rage followed by euphoria, so be warned.  Being creative is
 dangerous to some on the list.

Wow, I cant believe that someone else on this list is admitting to the
orgasm integrating a Definity with asterisk can give one. But it takes
so much foreplay and believe me its worth it. :)


 Try iaxmodem and hylafax.  Alex B did a very nice writeup on how to
 set that up so that it works very well.


 3. Now version 1.6 support Fax in a better way then 1.4. There is app_fax.c
 in the source code. Can someone please tell me what does this apps do?

 I assume you meant than not then, but I'm not sure if its a typo, you
 might just not know.
 app_fax.c does faxing. The question is what are YOU trying to do?


 4. Version 1.8 as two modes: SpanDSP mode and some other method, what is
 the different between then can someone please help?

 Whats the other mode?


 The best mode for T38 is probably Freeswitch or Callweaver.



 Thanks
 Sassy
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Re: [asterisk-users] FAX Issues

[C F]

Guys in my opinion this thread has been very productive. Which proves
one thing, as many people you are going to ask about faxing with
asterisk that many opinions you are going to get (maybe even add +1
opinions :P).
In the end it depends on your experience, hence I asked the OP to try
for himself.
Faxing is one of those things that for everyone something else works.
What works for you might and might not work for someone else.
To sum it up:
1. Faxing doesn't like lost packets.
2. That said, T.38 could work
3. Pure TDM (and by that I don't mean the literal definition of TDM
but POTS as well) is still the best.
4. Running that TDM thru asterisk is again open to different
experiences among users.
5. Using Asterisk as a fax endpoint (hylafax, app_fax take your pick)
should work in most cases if the handoff to asterisk is TDM and could
work if it's T.38



On Tue, Aug 9, 2011 at 9:28 PM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
 On Tue, Aug 9, 2011 at 8:31 PM, Lee Howard fax...@howardsilvan.com wrote:
 Steve Totaro wrote:

 On Tue, Aug 9, 2011 at 7:22 PM, Lee Howard fax...@howardsilvan.com
 wrote:


 Ryan McGuire wrote:


 Unless your network is under load and you are seeing dropped packets
 and high jitter, I would absolutely not do T.38. The cheapest and
 easiest approach that I have found is to buy yourself an FXS gateway
 and just make sure you are using ulaw.


 As SIP is usually running over UDP/IP it doesn't take much to produce
 dropped packets.  Dropped packets mean lost audio which means lost data
 and
 possible demodulation difficulties for the modems.  If you're in an
 environment where dropped UDP packets don't occur you're in a very rare
 scenario.


 I suppose you are talking about from the provider and not on the LAN?


 You certainly can (and usually will) have UDP packet loss on an uncontrolled
 LAN.

 Then you should be fired for not controlling your LAN.


 At Equinix in Ashburn VA, I have never had a dropped packet via the
 crossconnect from our cage to Level3's cage.  Sub ms pings.  Putting
 the primary PBX in Equinix and a 100meg speed for all VoIP calls in
 our out.  100meg DIA and 100meg layer 2 fiber to corporate.

 I have no reason to doubt your claims, but if this is true, then your
 arrangement there clearly mitigates the likelihood of UDP packet loss.
  Nevertheless, this arrangement is not something that the typical user who
 asks how do I fax over SIP/VoIP is going to have.  Without being very
 clear about the environment and explaining the pitfalls of not following
 your example exactly, you're not doing them any favors by encouraging them
 to attempt it in their environment.

 I don't care what you doubt or not.  I engineered it, Got the
 contracts signed, provisioned, setup BGP, got all that cleared with
 the steering committee.

 I do not assume anything about anyone and cannot define a typical
 user.  That is making as ass out of you and me.

 I encourage everyone to attempt and test everything!  Not put it in
 production without some reasonable testing.


 For every one user who I've ever heard from saying that they have reliable
 G.711 faxing over their SIP channels I've heard from a dozen who don't.

 Roger That.  That is why I advise clients it is not the best way to go.


 Thanks,

 Lee.

 He came in giving no details about his setup.  Now I see it is 1000 phones.

 I guess you don't care to read threads, and like to jump in the middle
 not knowing what you are talking about.

 I said that there is really no good way to handle faxes with Asterisk
 except with a T1, IAXmodem, and Hylafax.

 I bought fax for Asterisk when migrating to an all SIP world and it
 was a dismal failure and Digium didn't even have anyone that could
 explain any of the stats or problems I had.  I paid for it so I didn't
 have to screw around and finally I just took a refund and a big dent
 as far as the companies opinion of me.

 I wanted to try the 1.2 asterisk fork and that other project that came
 from the fact that Asterisk has deadlock issues and a whole cross
 platform rewrite has been done.

 Thanks,
 Steve Totaro

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Re: [asterisk-users] sip attacks

[C F]

How long ago was the last block from fail2ban?
What could be is that the attacker hasn't yet realized that he has
been blocked and is still trying, which although blocked by iptables
it is still coming down the line for attempted connections.

On Sun, Jul 31, 2011 at 7:04 PM, Dave George dgeo...@teletoneinc.com wrote:
 My asterisk server is getting bogged down every 5 minutes.  My ping time is
 going from 60ms to 800 ms and the call quality is bad.

 I have fail2ban running and I am using iptables.  I have two ip connections
 to the box.

 How can I tell if the poor performance is due to sip attacks?   I don't see
 any reg attempts in my asterisk cli.  I use to get frequent attacks but
 fail2ban seems to be taking care of that.

 See how ping time gets worst in a short space of time and server performance
 at the time:


 64 bytes from 4.2.2.1: icmp_seq=6 ttl=55 time=87.8 ms
 64 bytes from 4.2.2.1: icmp_seq=7 ttl=55 time=99.8 ms
 64 bytes from 4.2.2.1: icmp_seq=8 ttl=55 time=107 ms
 64 bytes from 4.2.2.1: icmp_seq=9 ttl=55 time=115 ms
 64 bytes from 4.2.2.1: icmp_seq=10 ttl=55 time=120 ms
 64 bytes from 4.2.2.1: icmp_seq=11 ttl=55 time=122 ms
 64 bytes from 4.2.2.1: icmp_seq=12 ttl=55 time=123 ms
 64 bytes from 4.2.2.1: icmp_seq=13 ttl=55 time=126 ms
 64 bytes from 4.2.2.1: icmp_seq=14 ttl=55 time=122 ms
 64 bytes from 4.2.2.1: icmp_seq=15 ttl=55 time=142 ms
 64 bytes from 4.2.2.1: icmp_seq=16 ttl=55 time=142 ms
 64 bytes from 4.2.2.1: icmp_seq=17 ttl=55 time=137 ms
 64 bytes from 4.2.2.1: icmp_seq=18 ttl=55 time=186 ms
 64 bytes from 4.2.2.1: icmp_seq=19 ttl=55 time=255 ms
 64 bytes from 4.2.2.1: icmp_seq=20 ttl=55 time=310 ms
 64 bytes from 4.2.2.1: icmp_seq=21 ttl=55 time=387 ms
 64 bytes from 4.2.2.1: icmp_seq=22 ttl=55 time=445 ms
 64 bytes from 4.2.2.1: icmp_seq=23 ttl=55 time=514 ms
 64 bytes from 4.2.2.1: icmp_seq=24 ttl=55 time=583 ms
 64 bytes from 4.2.2.1: icmp_seq=25 ttl=55 time=650 ms
 64 bytes from 4.2.2.1: icmp_seq=26 ttl=55 time=715 ms
 64 bytes from 4.2.2.1: icmp_seq=27 ttl=55 time=783 ms
 64 bytes from 4.2.2.1: icmp_seq=28 ttl=55 time=821 ms
 64 bytes from 4.2.2.1: icmp_seq=29 ttl=55 time=810 ms
 64 bytes from 4.2.2.1: icmp_seq=30 ttl=55 time=832 ms
 64 bytes from 4.2.2.1: icmp_seq=31 ttl=55 time=812 ms
 64 bytes from 4.2.2.1: icmp_seq=32 ttl=55 time=821 ms
 64 bytes from 4.2.2.1: icmp_seq=33 ttl=55 time=826 ms
 64 bytes from 4.2.2.1: icmp_seq=34 ttl=55 time=815 ms
 64 bytes from 4.2.2.1: icmp_seq=35 ttl=55 time=821 ms
 64 bytes from 4.2.2.1: icmp_seq=36 ttl=55 time=824 ms

 top - 19:02:38 up 4 days, 11:26,  4 users,  load average: 0.36, 0.75, 0.82
 Mem:   4051312k total,  1062964k used,  2988348k free,   167004k buffers
 Swap:  6094840k total,        0k used,  6094840k free,   680144k cached

  PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+  COMMAND
  4245 root      15   0  791m  86m  10m S 39.6  2.2   1192:32 asterisk
 18280 root      15   0  3812  600  516 S  2.0  0.0   0:59.00 pppoe
  2582 root      15   0  5912  628  504 S  0.3  0.0   2:02.19 syslogd
 18978 root      15   0 12744 1096  812 R  0.3  0.0   0:00.02 top
    1 root      15   0 10352  700  588 S  0.0  0.0   0:01.14 init
    2 root      RT  -5     0    0    0 S  0.0  0.0   0:00.01 migration/0
    3 root      34  19     0    0    0 S  0.0  0.0   0:31.90 ksoftirqd/0
    4 root      RT  -5     0    0    0 S  0.0  0.0   0:00.00 watchdog/0
    5 root      RT  -5     0    0    0 S  0.0  0.0   0:00.01 migration/1
    6 root      34  19     0    0    0 S  0.0  0.0   0:08.43 ksoftirqd/1
    7 root      RT  -5     0    0    0 S  0.0  0.0   0:00.00 watchdog/1
    8 root      RT  -5     0    0    0 S  0.0  0.0   0:00.13 migration/2
    9 root      34  19     0    0    0 S  0.0  0.0   2:40.56 ksoftirqd/2
   10 root      RT  -5     0    0    0 S  0.0  0.0   0:00.00 watchdog/2
   11 root      RT  -5     0    0    0 S  0.0  0.0   0:00.05 migration/3
   12 root      34  19     0    0    0 S  0.0  0.0   0:44.56 ksoftirqd/3
   13 root      RT  -5     0    0    0 S  0.0  0.0   0:00.00 watchdog/3
   14 root      10  -5     0    0    0 S  0.0  0.0   0:00.02 events/0
   15 root      10  -5     0    0    0 S  0.0  0.0   0:00.00 events/1
   16 root      10  -5     0    0    0 S  0.0  0.0   0:00.00 events/2
   17 root      10  -5     0    0    0 S  0.0  0.0   0:00.00 events/3
   18 root      10  -5     0    0    0 S  0.0  0.0   0:00.00 khelper
   55 root      10  -5     0    0    0 S  0.0  0.0   0:00.00 kthread
   62 root      10  -5     0    0    0 S  0.0  0.0   0:00.07 kblockd/0
   63 root      10  -5     0    0    0 S  0.0  0.0   0:00.01 kblockd/1
   64 root      10  -5     0    0    0 S  0.0  0.0   0:00.00 kblockd/2
   65 root      10  -5     0    0    0 S  0.0  0.0   0:00.00 kblockd/3
   66 root      17  -5     0    0    0 S  0.0  0.0   0:00.00 kacpid
  166 root      17  -5     0    0    0 S  0.0  0.0   0:00.00 cqueue/0
  167 root      18  -5     0    0    0 S  0.0  0.0   0:00.00 cqueue/1



 Dave



 --
 

Re: [asterisk-users] Security questions

[C F]

It's not bad but it wont prevent flooding your box with register
attempts and spoofing a user agent is trivia at best.

On Sat, Jul 23, 2011 at 9:09 PM, Flavio Miranda
flaviormira...@hotmail.com wrote:
 Hello everybody!

   I'd like to heard from those with more experience in Security if the
 following configuration is a good attempt to prevent hack:

 exten = CALLER,2,Set(header=${SIP_HEADER(User-Agent)})
 exten = CALLER,3,NoOp(Cabecalho ${header})
 exten = CALLER,4,GotoIf($[${header}= My User Agent]?6:7)

 Considering I have only one type of IP phone in my scenario.

 I know, somebody with another  IP phone will succeed in dial on my asterisk
 but I think it will limit at one only kind of IP phone.

 My question is , if there are some way to break it and use any kind of User
 Agent despite this configuratio.


 Att,

 Flavio Roberto Miranda
 MSN:flaviormira...@hotmail.com
 Skype: flaviormiranda
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Re: [asterisk-users] Securing Asterisk

[C F]

On Sat, Jul 23, 2011 at 1:38 PM, CDR vene...@gmail.com wrote:
 I beg to differ. Digium is hiding from the real world and somebody is

Because you have no clue how to secure a box its someone elses fault?

 going take the software and run with it. My customers lost in excess
 of $50.000 and cut my pay in half, because of hackers. The hackers

You deserved being fired all together. It was YOUR fault they hacked it.

 figured out how to scan every asterisk for weak passwords or open
 ports, and bang them real good. We need two things: a) disable in
 sip.conf the reply for INVITES that have wrong user information, and
 also, b) disable any response to any REGISTER packet altogether. Can
 somebody please write  patch? Or should we go broke trying to stop the
 flood of criminals coming from abroad?
 Federico

 On Sat, Jul 23, 2011 at 1:00 PM,
 asterisk-users-requ...@lists.digium.com wrote:
 Send asterisk-users mailing list submissions to
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 Today's Topics:

   1. Re: use dahdi for local terminal modem access? (Lyle Giese)
   2. dialplan pattern help (Armand Fumal)
   3. Re: Securing Asterisk - How to avoid sending, SIP/2.0 603
      Declined (Patrick Lists)
   4. Re: Securing Asterisk - How to avoid sending, SIP/2.0 603
      Declined (Paul Belanger)


 --

 Message: 1
 Date: Sat, 23 Jul 2011 09:29:26 -0500
 From: Lyle Giese l...@lcrcomputer.net
 Subject: Re: [asterisk-users] use dahdi for local terminal modem
        access?
 To: asterisk-users@lists.digium.com
 Message-ID: 4e2adac6.4010...@lcrcomputer.net
 Content-Type: text/plain; charset=ISO-8859-1; format=flowed


 On 07/22/11 22:47, William Stillwell wrote:
 Um, no VOIP involved here.

 Wrong.  What do you think Asterisk is?  Chopped meat?  It's a VoIP
 switch.  All traffic inside Asterisk is VoIP.


 I have an asterisk server with 2 23B+D PRI's

 I want to telnet/ssh into the asterisk server, and make an outbound call
 serial based modem/terminal connection (Like the 80/90's BBS Days).

 No TCP/IP or PPP or crazyness

 (ie, dialing into a Modem set to AA hooked to a Cisco Console Port)



 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
 boun...@lists.digium.com] On Behalf Of Lyle Giese
 Sent: Friday, July 22, 2011 8:07 PM
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] use dahdi for local terminal modem
 access?

 On 07/22/11 18:13, William Stillwell wrote:
 I have some terminals that have phone lines.

 One of my tech had an idea of using IAXmodem or something similar to
 use
 existing PRI/DAHDI Trucks for dial out via the asterisk/Linux
 console.

 Anybody ever heard of doing this?

 I would think maybe would use iaxmodem maybe and a shell terminal
 app?

 (basically I'm dialing into a remote access device that uses a pots
 like
 for remote administration, and don't want to string a channel bank
 off
 my asterisk box, and a hook to a modem)



 --

 Depends on your expectation.  Because of compression in the codecs, it
 will be hard to get fast dialup.  If you mean ssh or telnet, it might
 work.  If you mean vnc or RDP over this, you may not get enough usable
 bandwidth to do that.

 Given this, I have in an emergency dialed into a RAS server via a VoIP
 line. My laptop connected at 14,400bps.  All I needed to do was telnet
 into an APC masterswitch to toggle power on one outlet.  It worked.

 I was surprised at getting a 14,400bps connect.  I was not expecting
 that high and really did not need that high.  300 baud probably would
 have been fast enough to telnet into an APC masterswitch.

 Lyle Giese
 LCR Computer Services, Inc.

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Re: [asterisk-users] 10.0.0 better than 2.0.0?

[C F]

Since this change I started measuring temperature in Rankine. Its now
592.67 degrees here (south NJ).

On Fri, Jul 22, 2011 at 3:44 PM, Tony Mountifield t...@mountifield.org wrote:
 I read Kevin's piece in asterisk-announce about the new numbering scheme,
 and saw in svn-commits some tagging of 10.0.0-beta1.

 Perhaps I'm thick (I hope not!), but I really can't see why calling the
 next version 10.0.0 is any better than calling it 2.0.0!

 I'm surprised not to have seen ANY talk in asterisk-users or aserisk-dev
 about it, since the announcement.

 Cheers
 Tony
 --
 Tony Mountifield
 Work: t...@softins.co.uk - http://www.softins.co.uk
 Play: t...@mountifield.org - http://tony.mountifield.org

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Re: [asterisk-users] FAX with SIP

[C F]

Short answer is: dont use it. For the long answer wait for others to
answer that.

On Mon, Jul 18, 2011 at 7:20 AM, Eduardo Carpes car...@bsd.com.br wrote:
 Hello guys
 I need some help to do works FAX using SIP, anybody know the secret to
 this? Have asterisk 1.6.
 Thanks!!

 --
 Enviado do meu celular

 Eduardo Carpes
 E-mail: car...@bsd.com.br
 www.freebsd.org

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Re: [asterisk-users] REALY strange issue with making calls biside 2 phones

[C F]

what does sip show peers say?

On Tue, Jul 12, 2011 at 12:40 PM, Matiss Jekabsons mat...@jekabsons.lv wrote:
 Thats my issue, i hope someone could suggest something:

 Phone A - Phone B



 == Using SIP RTP CoS mark 5

    -- Executing [01@default:1] Dial(SIP/00-0076, SIP/01)
 in new stack

  == Using SIP RTP CoS mark 5

    -- Called 01

    -- SIP/01-0077 is ringing

    -- SIP/01-0077 answered SIP/00-0076

    -- Locally bridging SIP/00-0076 and SIP/01-0077

  == Spawn extension (default, 01, 1) exited non-zero on
 'SIP/00-0076'







 Phone B - phone A



  == Using SIP RTP CoS mark 5

    -- Executing [00@default:1] Dial(SIP/01-0078, SIP/00)
 in new stack

 [Jul 12 19:08:35] WARNING[2965]: app_dial.c:2039 dial_exec_full: Unable to
 create channel of type 'SIP' (cause 20 - Unknown)

  == Everyone is busy/congested at this time (1:0/0/1)

    -- Executing [00@default:2] Hangup(SIP/01-0078, ) in new
 stack

  == Spawn extension (default, 00, 2) exited non-zero on
 'SIP/01-0078'



 --
 --
 Best regards
 Matiss Jekabsons
 Procerto Ltd.




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Re: [asterisk-users] Conference feature

[C F]

Does asterisk support it?

On Sun, Jun 26, 2011 at 9:25 PM, Rafael dos Santos Saraiva
rafaels...@gmail.com wrote:
 Hi
 How to create the conference feature in Asterisk?
 Thank's
 Att,
 Rafael Saraiva

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Re: [asterisk-users] DIGIUM PRI CARDS REQUIRE

[C F]

Tzafrir, Whats up with this 1.2 vs 1.8 signature?


On Thu, Jun 16, 2011 at 3:38 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
 Hi,

 I hope this is not rude of my part. I normally avoid answering mails
 that relate in such way commercially to hardware.

 This list is non-commercial If you want to ask questions of commercial
 nature, please use Asterisk-biz:

  http://lists.digium.com/mailman/listinfo/asterisk-biz

 Please follow up on this thread in privat email and not on-list.

 (For the record: this mail was sent after on-list Mahesh Katta's reply).

 Regards,


 Oh, and: you should avoid using Asterisk 1.2 on a new installation.
 Please use Asterisk 1.8 ;-)

 --
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 icq#16849755              jabber:tzafrir.co...@xorcom.com
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Re: [asterisk-users] sig_pri.c:985 pri_find_dchan: Span 1: No D-chanannel anyway!

[C F]

You know I'm a flight engineer but non of the airlines wanted to hire
me because other than the self proclaimed title I have no clue how to
operate or maintain an aircraft.
The dictionary is probably wrong you should patch libpri

On Tue, Jun 14, 2011 at 11:43 AM, bilal ghayyad bilmar...@yahoo.com wrote:
 The provider came to the site and they have a gateway, they connected the 
 gateway to the E1 cable, and we called the number and it answered !!!

 So it is no more provider issue.

 I start beleive, it is a bug maybe in the dahdi and libpri. I tried all the 
 possibilities, but no luck.

 WHAT COULD BE?


 Regards
 Bilal
 --

 bilal ghayyad wrote:
  There is a Yellow Alarm, so what it could be the
 problem?

 Experience says you need to call your provider.

 Doug


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Re: [asterisk-users] sig_pri.c:985 pri_find_dchan: Span 1: No D-channels available! Using Primary channel as D-channel anyway!

[C F]

Its probably not a bug so don't apply this patch. No D-Channel means
it cant sync up. It could be related to anything but the least likely
is that its a bug in libpri or dahdi.
Just go thru your configs, check and double check the cabling.


On Tue, Jun 14, 2011 at 7:27 PM, bilal ghayyad bilmar...@yahoo.com wrote:
 Dear;

 Thanks a lot for guiding me.

 Is it possible that the installation libpri-1.4.11.5 newer than the 
 libpri-1.4.11.5-patch?

 Well, when I typed (note: I am trying to apply the libpri-1.4.11.5-patch for 
 the libpri-1.4.11.5):

 libpri-1.4.11.5# patch -p0 -i libpri-1.4.11.5-patch

 It gave me that patched detected as shown below (example of one file, and I 
 got same for other files):

 patching file pri.c
 Reversed (or previously applied) patch detected!  Assume -R?

 And when I reached to the q931, then it failed to save and apply the hunk !!


 patching file q931.c
 Hunk #1 succeeded at 601 (offset 280 lines).
 Hunk #2 FAILED at 4945.
 Hunk #3 FAILED at 4957.
 Hunk #4 FAILED at 4973.
 Hunk #5 FAILED at 5001.
 Hunk #6 FAILED at 5303.
 Hunk #7 FAILED at 5319.
 Hunk #8 FAILED at 5817.
 Hunk #9 FAILED at 5830.
 Hunk #10 FAILED at 6085.
 Hunk #11 FAILED at 6236.
 Hunk #12 FAILED at 6379.
 Hunk #13 FAILED at 6447.
 Hunk #14 FAILED at 6635.
 Hunk #15 FAILED at 7287.
 Hunk #16 FAILED at 7320.
 Hunk #17 FAILED at 7332.
 Hunk #18 FAILED at 7426.
 Hunk #19 FAILED at 7991.
 Hunk #20 FAILED at 8029.
 Hunk #21 FAILED at 8077.
 Hunk #22 FAILED at 8096.
 21 out of 22 hunks FAILED -- saving rejects to file q931.c.rej

 What this mean? The installation libpri-1.4.11.5 already contains the 
 libpri-1.4.11.5-patch and the install libpri-1.4.11.5 already newer than the 
 libpri-1.4.11.5-patch? So it failed to apply the changes to the q931 file?

 As I see that I am facing a problem with applying the patch to the q931 while 
 at the same time the D-channel is related to the q931, so it could seriously 
 a bug?

 I am now confused, not able to know if the installation newer than the patch 
 or I am missing something in doing the patch?

 In case the installation newer than the patch, so I will resolve my problem? 
 Shall I come back for previous version or I have to go for beta version :( !!

 Please help.

 Regards
 Bilal

 


 http://www.linuxtutorialblog.com/post/introduction-using-diff-and-patch-tutorial

  -Original Message-
  From: asterisk-users-boun...@lists.digium.com
  [mailto:asterisk-users-boun...@lists.digium.com]
 On Behalf Of
  bilal ghayyad
  Sent: Tuesday, June 14, 2011 2:47 PM
  To: asterisk-users@lists.digium.com
  Subject: Re: [asterisk-users] sig_pri.c:985
 pri_find_dchan:
  Span 1: D-chanannel anyway!
 
  Dears;
 
  To patch libpri: I just place the patch file in the
 libpri
  source directory and then I run make and make
 install?
 
  Or I need to compile the dahdi and asterisk also?
 
  If the problem stayed, do I have to go for previous
 libpri
  version? Or for previous dahdi version and asterisk
 version?
 
  Regards
  Bilal
 
  ---
 
   bilal ghayyad wrote:
But I am afraid it is a bug because I read
 something
   this in the below
  
   This bug is referring to Zaptel, not dahdi.
  
   If things were working fine, and you haven't made
 any
   recent changes, in
   my experience it's always been provider (99%) or
 cable (1%)
   issues when
   I've lost my D channel.
  
   Doug
  
   --
  
  
   You know I'm a flight engineer but non of the
 airlines
   wanted to hire
   me because other than the self proclaimed title I
 have no
   clue how to
   operate or maintain an aircraft.
   The dictionary is probably wrong you should patch
 libpri
  
   On Tue, Jun 14, 2011 at 11:43 AM, bilal ghayyad
  bilmar...@yahoo.com
   wrote:
The provider came to the site and they have
 a gateway,
   they connected the gateway to the E1 cable, and
 we called
   the number and it answered !!!
   
So it is no more provider issue.
   
I start beleive, it is a bug maybe in the
 dahdi and
   libpri. I tried all the possibilities, but no
 luck.
   
WHAT COULD BE?
   
   
Regards
Bilal
--
 


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Re: [asterisk-users] Configuring Cisco Phones to register on Asterisk: The configuration files

[C F]

You gotto love this this guy. You can almost predict what his next question is.

On Sun, Jun 12, 2011 at 5:16 PM, bilal ghayyad bilmar...@yahoo.com wrote:
 Hi All;

 I need to create the needed files for the Cisco Phones to be placed in the 
 TFTP server to be able to  register on Asterisk.

 I need a help in the following please:

 1) Regarding to the file: SIPDefault.cnf, The proxy1_address is the IP 
 address of Asterisk?

 2) Regarding to the file: SIPmacaddress.cnf, if I need to let it appear at 
 the Phone (the first line for example) the extension, so this will be the 
 line1_name? For example, if I need the extension to be 700 then I set 
 line1_name : 700 ? And if I need line 2 to be 701 then I set line2_name : 701?

 3) About the xmlDefault.CNF.XML, it is mention that I have to place the IP 
 address of Asterisk, but where? What is the format to write the Asterisk IP 
 address in this file?

 4) About the dialplan.xml file, what the below means?

 TEMPLATE MATCH=* Timeout=5/ !-- Anything else --

 Any help?
 Regards
 Bilal

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Re: [asterisk-users] Configuring Cisco Phones to register on Asterisk: The configuration files

[C F]

ROFL

On Sun, Jun 12, 2011 at 6:27 PM, Alex Balashov
abalas...@evaristesys.com wrote:
 And that means he can be replaced with a small shell script.

 --
 Alex Balashov - Principal
 Evariste Systems LLC
 260 Peachtree Street NW
 Suite 2200
 Atlanta, GA 30303
 Tel: +1-678-954-0670
 Fax: +1-404-961-1892
 Web: http://www.evaristesys.com/

 On Jun 12, 2011, at 5:59 PM, C F shma...@gmail.com wrote:

 You gotto love this this guy. You can almost predict what his next question 
 is.

 On Sun, Jun 12, 2011 at 5:16 PM, bilal ghayyad bilmar...@yahoo.com wrote:
 Hi All;

 I need to create the needed files for the Cisco Phones to be placed in the 
 TFTP server to be able to  register on Asterisk.

 I need a help in the following please:

 1) Regarding to the file: SIPDefault.cnf, The proxy1_address is the IP 
 address of Asterisk?

 2) Regarding to the file: SIPmacaddress.cnf, if I need to let it appear 
 at the Phone (the first line for example) the extension, so this will be 
 the line1_name? For example, if I need the extension to be 700 then I set 
 line1_name : 700 ? And if I need line 2 to be 701 then I set line2_name : 
 701?

 3) About the xmlDefault.CNF.XML, it is mention that I have to place the IP 
 address of Asterisk, but where? What is the format to write the Asterisk IP 
 address in this file?

 4) About the dialplan.xml file, what the below means?

 TEMPLATE MATCH=* Timeout=5/ !-- Anything else --

 Any help?
 Regards
 Bilal

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Re: [asterisk-users] call files .vbs

[C F]

I'm the original author of said VB Script.
Steve is right, I had lots of errors - related to the fact that
asterisk watches it too closely and reads the files even before they
are complete - and have since updated it that it first dumps it to a
temp directory, then use a bash script on the linux machine that moves
all files from the temp directory to the call directory using plink.
Both pscp and plink are windoz programs that utilize ssh for their
functions. Pscp xfers files, and plink executes any remote commands.
In the newer version pscp in the VB Script dumps it to
/root/calltemps/ and /root/mvcallfiles.sh moves the files from
/root/calltemps/* to /var/spool/asterisk/outgoing/
change this line:
strcmd=C:\pscp -pw password c:\direcotry\strcnt\*
root@asterisk:/var/spool/asterisk/outgoing
to:
strcmd=C:\pscp -pw password c:\directory\strcnt\*
root@asterisk:/root/calltemps
make sure the dir exists
then add:
Set objShell2 = CreateObject(WScript.Shell)
strcmd2=C:\plink -pw password root@asterisk /root/mvcallfiles.sh
objShell2.Run strcmd2
/root/movcallfiles.sh:
#/bin/bash

mv /root/calltemps/* /var/spool/asterisk/outgoing/

Hope this helps.






On Sun, May 22, 2011 at 8:55 PM, Steve Edwards
asterisk@sedwards.com wrote:
 On Sun, 22 May 2011, Thomas Perron wrote:

 Also, it seems like the pscp function is the way that I can tie together
 the vb script with the logic of the Asterisk call files learning
 curve!!

 pscp is a program, not a function. Part of or related to putty as I
 remember.

 Not a good idea. One of the 'bugaboos' of call files is that you are
 supposed to create the files in a temporary directory and move them into the
 spool directory.

 Also, you will have limited error detection ability if you are only dumping
 files 'willy-nilly.'

 Much better to do it all on the Asterisk host.

 --
 Thanks in advance,
 -
 Steve Edwards       sedwa...@sedwards.com      Voice: +1-760-468-3867 PST
 Newline                                              Fax: +1-760-731-3000

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Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not

[C F]

+1

/@
\ \
  ___ \
 (__O)  \
(@)  \
(@)   \
 (__o)_\
   \\




On Tue, May 10, 2011 at 4:23 PM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
 I'll keep this brief because I don't want to come across like any more of an
 a$$ than I absolutely have to, especially since I know I've blown my stack
 before.

 Gentlemen (and Ladies, if you're out there),

 If someone gives you advice on this list, and ESPECIALLY if they give you
 advice offlist, have the courtesy to (AT THE LEAST) to let them know when/if
 you get your question answered or your problem solved.

 As many people point out, on community supported mailing lists and forums
 around the world, these user lists are comprised of people who are giving
 their time freely to help others learn about the software the list is about.
 Sometimes those lists are about software that is quite useful in a
 commercial setting, perhaps even very much in demand, like Asterisk. Now,
 you should always appreciate when you get assistance from people on user
 lists, but when you're asking for help on a list like this one, (where I'd
 say 80% of the participants on the list are professionals who earn their
 living by selling their knowledge of how to install, configure, and maintain
 a server application like Asterisk) it would be extremely appreciated if you
 show some courtesy to the individual(s) who assisted you for free. I've had
 several individuals contact me offlist (without being given permission
 first, which is first and foremost bad form) and ask for my assistance with
 configuring a feature, troubleshooting an issue, and once I got an email
 that said something along the lines of:
 I saw a post on the list where you said you could accomplish
 insertNiftyFeatureThatDidNotPreviouslyExistHere Tell me how to do it
 I'm sure many of you have been the recipient of more than your fair share of
 emails offlist asking for help, and I'm sure a great number of you try to
 offer assistance. What is bothering me is the fact there seems to be a new
 trend forming, wherein I don't get a repsonse from the person I tried to
 help, even when I can feel confident in saying that I know I gave them the
 piece of information they needed in order to answer their question and
 accomplish the goal of making Asterisk perform the way they wanted.

 Has anyone else noticed this trend?

 Those of you who are making the requests, is there a reason why you don't
 feel the need to be courteous and at least say, Hey that advice worked,
 everything's working now?

 Next time you ask for help, especially when it's offlist (and even MORE SO
 when you're contacting someone you weren't invited to contact offlist), I
 want you to remember that the person you're contacting usually gets paid for
 their time as an Asterisk professional, and that they're helping you for
 free. Hell, if you want to get down to brass tacks about it, thatr person
 who is taking the time to try and help you is increasing his or her own
 professional competition..


 that's all...nothing super rude, but I had to get that one out there I
 get annoyed when I answer about 12-13 questions (all in separate emails,
 mind you) from someone, and then I never get even find out if I was
 successful in helping them
 --
 Sherwood McGowan
 Telecommunications and VOIP Consultant


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Re: [asterisk-users] Retrieving sound files from DB as opposed to filesystem

[C F]

On Tue, May 3, 2011 at 1:09 AM, A E [Gmail] all.efor...@gmail.com wrote:
 On Mon, May 2, 2011 at 9:45 PM, C F shma...@gmail.com wrote:

 Just from my experience with different DBs, stay away from BLOB data
 types as much as possible.

 Hi CF,
 any particular reason why? I've had a good experience with it, in fact
 that's recommended by DB developers when it's a case of small files. They
 say only larger files greater than 500K-1MB should be stored on the
 filesystem using filestream or similar etc.
 Although at this point, this might be a moot point, as so far no one's been
 able to suggest a way to be able to stream the content of the BLOB field to
 Asterisk over the AGI connection into the current channel, such that
 Asterisk can just play it on the fly. We'll have to just go with getting the
 file to the requesting * server and then play it

Like I said it's based on experience, I can try hacking my brain and
come up with particular reasons that can be written down, however I
don't really have the time for that right now.
In any event, my experience has only been with larger files (1MB up to
25MB). I never tried smaller ones, and since I just don't use it
anymore I doubt I will find out any time soon.

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Re: [asterisk-users] best current version and motherboard/CPU compatibilities

[C F]

On Mon, May 2, 2011 at 11:46 PM, || dave cantera Mobile
david.cant...@ibsonecall.com wrote:
 I've been away from asterisk for a while since 1.4.16 and only installed 1.6
 once to run a test... can someone recommend what the best version to install
 is and the recommended CPU/motherboard for an * box these days? I'm just
 running about 20 handsets and 4-8 lines with POTS  SIP mix.

 I remember there were some issues with bios a while back and a TDM card was
 required for timing conferencing, etc... are these requirements still an
 issue?

 I want to setup another * box and was wondering which CPU/motherboard to
 select...
 thanks,
 daveC


I like my machine, I have been using it for some 5 years now:

root@pbx:~# cat /proc/cpuinfo
processor   : 0
vendor_id   : AuthenticAMD
cpu family  : 5
model   : 8
model name  : AMD-K6(tm) 3D processor
stepping: 0
cpu MHz : 350.801
cache size  : 64 KB
fdiv_bug: no
hlt_bug : no
f00f_bug: no
coma_bug: no
fpu : yes
fpu_exception   : yes
cpuid level : 1
wp  : yes
flags   : fpu vme de pse tsc msr mce cx8 mmx syscall 3dnow
bogomips: 699.59


root@pbx:~# cat /proc/meminfo
total:used:free:  shared: buffers:  cached:
Mem:  129593344 126328832  32645120 12873728  8675328
Swap: 518152192  5279744 512872448
MemTotal:   126556 kB
MemFree:  3188 kB
MemShared:   0 kB
Buffers: 12572 kB
Cached:   6900 kB
SwapCached:   1572 kB
Active:   5072 kB
Inactive:15996 kB
HighTotal:   0 kB
HighFree:0 kB
LowTotal:   126556 kB
LowFree:  3188 kB
SwapTotal:  506008 kB
SwapFree:   500852 kB


pbx*CLI show version
Asterisk 1.2.13 built by root @ pbx on a i586 running Linux on
2006-11-14 04:10:38 UTC
pbx*CLI show  uptime
System uptime: 5 weeks, 1 day, 22 hours, 2 minutes, 35 seconds


Runs fine and is only up for that short because the last power outage was then.

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Re: [asterisk-users] Retrieving sound files from DB as opposed to filesystem

[C F]

Just from my experience with different DBs, stay away from BLOB data
types as much as possible.

On Mon, May 2, 2011 at 3:15 AM, A E [Gmail] all.efor...@gmail.com wrote:
 Hello All,
 Probably a silly question, but we're wondering if people have had any
 experience and have data to demonstrate if the performance of the Asterisk
 system might suffer in terms of latency etc. if we're to have it retrieve
 sound files from a database using odbc as opposed to storing them locally on
 the filesystem. Note, these are not prompts...these are sound files that are
 being created through a web-app and being stored in the DB as BLOB or
 similar datatype that's good/efficient to store audio/video files in a DB.
 We need these be made available through the asterisk system to play over the
 phone. Although the DB uses a SAN, the Asterisk System has no connectivity
 to the SAN but is connected on the same physical ethernet switch with a
 multi-Gbps backplane.
 The way the system is being designed, it's possible for us to end up with
 000s of these sound files stored in the DB, not to mention several asterisk
 systems in a pool/cluster/farm requesting these files, so using the local
 filesystem might not be scalable or efficient.
 Any advice/comments/suggestions welcome :)


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Re: [asterisk-users] PAP2T auto answer?

[C F]

The answer function on an analog line is accomplished by going off
hook. Unless the line is controlled by an automated device (like
answering machine) someone has to physically take the device off hook
to answer it. The ATA has no way to do it as all it gives is the FXS
signalling.
What exactly are you trying to accomplish?
Vikingeleoctronics makes a door box (E20 iirc) that is powered by an
analog line and can do auto answer when it gets the first ring.

On 4/25/11, Mike Diehl mdi...@diehlnet.com wrote:
 Hi all,

 Is it possible to send a SIP header to a PAP2T or SPA and cause the
 device
 to automatically answer?  I can do this with my Polycom phones and would
 like
 to do it with my ATA's.

 Any ideas?

 --

 Take care and have fun,
 Mike Diehl.

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Re: [asterisk-users] Asterisk as a Condo door opener/intercom

[C F]

On Mon, Apr 11, 2011 at 7:21 PM, Don Kelly d...@donkelly.biz wrote:
 Continuing top posting...

 The same argument could be made for any commercial solution. Why use
 Asterisk when we could throw $4,000 at our problem for a commercial
 solution?

Really


 I'd like to have a solution that would have the features you suggest for
 $400.

Doesnt exist on planet earth not even with Asterisk. The closest
you'll get to your mentioned price is a commercial solution.


 --Don


 On Behalf Of C F
 Sent: Monday, April 11, 2011 11:43 AM

 Search the lists. Some hints:
 Viking electronics makes a door box that connects to any analog line
 (IIRC e-20).
 They also make a DTMF keypad that integrates in series with any analog
 line. They might also make a door box with a DTMF keypad on it.
 Sandman makes a relay that will get energized when there is a ring on
 the line which could be used to unlock the door.

 However, why would you use asterisk? Using asterisk for the sole
 purpose of MDU entry system is like using windows for asterisk, it
 works but why?
 Go for the commercial solutions, it comes with a geziilion options for
 your setup one of them the ability of chosing an apartment, another
 add key fobs, another one is the ability of using a code for the
 residence (not guests) to unlock the door. Also the interface with
 asterisk you will have to build one from scratch. The commercial
 solutions have em built in.

 On 4/10/11, Bruce B bruceb...@gmail.com wrote:
 Hi Everyone,

 Looking to replace a condo intercom system. Apparently the current one
 taps
 into the lines and dials phone numbers but needs to be changed as it's
 faulty.

 I will probably still use the same analogue dialing and back it up with a
 VoIP line and use the current cabling that is in place. But as for as the
 door opening function goes, I am not sure how to interface and how open
 these modules are usually built.

 I would appreciate it if someone with experience can throw in some
 pointers
 as to what I might be facing and what challenges I have to solve to
 replace
 this with a nice Asterisk system.

 Thanks,


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Re: [asterisk-users] Asterisk as a Condo door opener/intercom

[C F]

Search the lists. Some hints:
Viking electronics makes a door box that connects to any analog line
(IIRC e-20).
They also make a DTMF keypad that integrates in series with any analog
line. They might also make a door box with a DTMF keypad on it.
Sandman makes a relay that will get energized when there is a ring on
the line which could be used to unlock the door.

However, why would you use asterisk? Using asterisk for the sole
purpose of MDU entry system is like using windows for asterisk, it
works but why?
Go for the commercial solutions, it comes with a geziilion options for
your setup one of them the ability of chosing an apartment, another
add key fobs, another one is the ability of using a code for the
residence (not guests) to unlock the door. Also the interface with
asterisk you will have to build one from scratch. The commercial
solutions have em built in.

On 4/10/11, Bruce B bruceb...@gmail.com wrote:
 Hi Everyone,

 Looking to replace a condo intercom system. Apparently the current one taps
 into the lines and dials phone numbers but needs to be changed as it's
 faulty.

 I will probably still use the same analogue dialing and back it up with a
 VoIP line and use the current cabling that is in place. But as for as the
 door opening function goes, I am not sure how to interface and how open
 these modules are usually built.

 I would appreciate it if someone with experience can throw in some pointers
 as to what I might be facing and what challenges I have to solve to replace
 this with a nice Asterisk system.

 Thanks,


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Re: [asterisk-users] Removing Polycom Transfer Softkey

[C F]

Sorry, for some reason I misread it as the forward feature.


On Sun, Mar 27, 2011 at 9:16 PM, Mark Murawski
markm-li...@intellasoft.net wrote:
 From the polycom pdf:

 divert.fwd.x.enabled
 If set to 1, the user will be able to enable universal call
 forwarding through the soft key menu.

 This sounds like it turns on and turns off the call forwarding feature on
 the phone.  I can try it out Monday, but I don't see where it has any
 relation to transfer (both attended and blind).




 On 03/27/2011 08:43 PM, C F wrote:

 In phone.cfg set the following line to
 divert.fwd.1.enabled=0
 from:
 divert.fwd.1.enabled=1
 For more info check page 323:

 http://supportdocs.polycom.com/PolycomService/support/global/documents/support/setup_maintenance/products/voice/spip_ssip_vvx_Admin_Guide_SIP_3_2_2_eng.pdf



 On Fri, Mar 25, 2011 at 6:33 PM, Mark Murawski
 markm-li...@intellasoft.net  wrote:

 Sorry for the crosspost.  This was supposed to be on -users


 I know some of you are polycom gurus...

 Anyone know how to remove transfer from a polycom 33x phone?  We've set
 allowtransfer=no, but we would like to remove a polycom soft key as well.

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Re: [asterisk-users] Removing Polycom Transfer Softkey

[C F]

Look at page 311 in that manual
If you disable the soft keys and then reassign the hard key it should
- at least in theory - be possible to accomplish.


On Tue, Mar 29, 2011 at 11:46 AM, C F shma...@gmail.com wrote:
 Sorry, for some reason I misread it as the forward feature.


 On Sun, Mar 27, 2011 at 9:16 PM, Mark Murawski
 markm-li...@intellasoft.net wrote:
 From the polycom pdf:

 divert.fwd.x.enabled
 If set to 1, the user will be able to enable universal call
 forwarding through the soft key menu.

 This sounds like it turns on and turns off the call forwarding feature on
 the phone.  I can try it out Monday, but I don't see where it has any
 relation to transfer (both attended and blind).




 On 03/27/2011 08:43 PM, C F wrote:

 In phone.cfg set the following line to
 divert.fwd.1.enabled=0
 from:
 divert.fwd.1.enabled=1
 For more info check page 323:

 http://supportdocs.polycom.com/PolycomService/support/global/documents/support/setup_maintenance/products/voice/spip_ssip_vvx_Admin_Guide_SIP_3_2_2_eng.pdf



 On Fri, Mar 25, 2011 at 6:33 PM, Mark Murawski
 markm-li...@intellasoft.net  wrote:

 Sorry for the crosspost.  This was supposed to be on -users


 I know some of you are polycom gurus...

 Anyone know how to remove transfer from a polycom 33x phone?  We've set
 allowtransfer=no, but we would like to remove a polycom soft key as well.

 --

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Re: [asterisk-users] Removing Polycom Transfer Softkey

[C F]

In phone.cfg set the following line to
divert.fwd.1.enabled=0
from:
divert.fwd.1.enabled=1
For more info check page 323:
http://supportdocs.polycom.com/PolycomService/support/global/documents/support/setup_maintenance/products/voice/spip_ssip_vvx_Admin_Guide_SIP_3_2_2_eng.pdf



On Fri, Mar 25, 2011 at 6:33 PM, Mark Murawski
markm-li...@intellasoft.net wrote:
 Sorry for the crosspost.  This was supposed to be on -users


 I know some of you are polycom gurus...

 Anyone know how to remove transfer from a polycom 33x phone?  We've set
 allowtransfer=no, but we would like to remove a polycom soft key as well.

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Re: [asterisk-users] What is the most stable version of asterisk?

[C F]

1.4 is the new flavor for my new deployments, but I definitely have
more (way more, like 1:8) 1.2 systems in production.

On Fri, Mar 25, 2011 at 10:32 AM, Douglas Mortensen
d...@impalanetworks.com wrote:
 Do you have the same ratio of deployments using 1.4 as you do with 1.2? What 
 about 1.6 or 1.8? I simply question how accurate a comparison can be made 
 when one is comparing 50 1.2 deployments to 5 1.4 deployments? I'm sure it 
 says something, and I do appreciate the feedback.

 -
 Doug Mortensen
 Network Consultant
 Impala Networks
 P: 505.327.7300
 .

 -Original Message-
 From: C F [mailto:shma...@gmail.com]
 Sent: Thursday, March 24, 2011 8:14 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] What is the most stable version of asterisk?

 I use mainly 1.2 with great success, mostly restarts are due to power outages.
 I recently started to upgrade to 1.4, so far so good. Too early to say, the 
 longest running 1.4 is only 234 days. While I have had 900+ days with 1.2



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Re: [asterisk-users] What is the most stable version of asterisk?

[C F]

I use mainly 1.2 with great success, mostly restarts are due to power outages.
I recently started to upgrade to 1.4, so far so good. Too early to
say, the longest running 1.4 is only 234 days. While I have had 900+
days with 1.2

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Re: [asterisk-users] Asterisk

[C F]

I like the subject.

On Sun, Mar 20, 2011 at 4:03 AM, Kaushal Shriyan
kaushalshri...@gmail.com wrote:
 Hi,
 I have couple of questions regarding Asterisk.
 a) Does it has Automated Dialing Feature like dialing 1000 and 1000 of phone
 numbers?
 b) Does it Support VoiceXML ?
 c) What PRI Card is recommended for using Asterisk ?
 Thanks
 Kaushal




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Re: [asterisk-users] Metaswitch to Asterisk problems

[C F]

Make sure you get some DNIS, on the meta you might have to play around
to get it, it might come in in the form of some sip headers or as
asterisk expects it as an extension. In any event, once asterisk knows
which extension (DID) it belongs to just send it to VM.


On Thu, Mar 10, 2011 at 10:02 AM, Chris Ledford
chris.ledf...@cnxntech.com wrote:
 I am setting up VM off Metaswitch due to a problem with Metaswitch VM. I
 have a couple days to prove this works and I need a little assist please.



 I am using TRIXBOX 2.6.2.5 and have the Meta SIP trunk up. I have extensions
 built that can talk to each other. I took a trace on the TRIXBOX that shows
 when I dial my test phone on Metaswitch it goes to VM after a couple rings
 and the call goes to my TRIXBOX and asterisk plays a RTP message saying that
 the number cannot be contacted.



 I don’t understand how the TN on Metaswitch translates to the TRIXBOX VM
 account.



 Can I please get a response on or off forum for some assistance in what I am
 doing wrong or if there is a good post or website with a guide on this
 config..



 I have looked at several online over that last couple of weeks and they are
 dated and not running TRIXBOX 2.6.2.5, so trying to conform what I am
 reading to my config is confusing.



 Thanks in advance.



 V/r



 Chris Ledford

 CCNA/CCSP/CCNP Voice

 Comptia A+/Net+/Linux+/Sec+

 EWC/CTTC(sw) USN
 T3 Engineer
 http://navy.togetherweserved.com/profile/13552

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Re: [asterisk-users] How do I find a phone numbers issued by Rogers?

[C F]

Call them.

On Wed, Mar 2, 2011 at 8:17 PM, Robert Augustyn
robert.augus...@linqone.com wrote:
 Hi,

 Is there a way of finding out what block of phone numbers were issued to
 Roger’s business customers in my end of the woods?

 Thanks,

 Sincerely,



 Robert Augustyn

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Re: [asterisk-users] Problem installing FXS module in old digium 4 channel tdm card

[C F]

This is the closest thing I was able to find in my wctdm.c file:
if ((blah  0xf) == 2) {
/* ProSLIC 3215, not a 3210 */
wc-flags[card] |= FLAG_3215;
}
If I take out the 2 first lines I get errors when compling.


On Tue, Feb 22, 2011 at 11:43 PM, Shaun Ruffell sruff...@digium.com wrote:
 On Tue, Feb 22, 2011 at 11:00:22PM -0500, C F wrote:
 On Mon, Feb 21, 2011 at 11:23 PM, Shaun Ruffell sruff...@digium.com wrote:
  On 2/21/11 4:46 PM, C F wrote:
  I just installed an FXS module onto a 4 channel tdm thats about 5
  years old and it wont work. Running dmesg I can see the following
  error:
 

 [snip]

   ! Init Indirect Registers UNSUCCESSFULLY.
  Indirect Registers failed verification.
  Module 0: FAILED FXS (FCC)
  Module 1: Installed -- AUTO FXO (FCC mode)
  Module 2: Installed -- AUTO FXO (FCC mode)
  Module 3: Installed -- AUTO FXO (FCC mode)
  Found a Wildcard TDM: Wildcard TDM400P REV E/F (3 modules)
 
  Does this have to do with the fact that the module is way newer than the
  card?
 
 
  Not having much direct experience with the wctdm.c driver, that would be my
  guess. You might be able to edit the wctdm_proslic_insane() function to
  force the FLAG_3215 on for the card and see if that gives you a different
  result.
 

 How/Where would I do that?


 Around line 1297 of drivers/dahdi/wctdm.c you could change:

  if (wctdm_getreg(wc, card, 1)  0x80)
                /* ProSLIC 3215, not a 3210 */
                wc-flags[card] |= FLAG_3215;

  to

  wc-flags[card] |= FLAG_3215;

 and just skip the read of register 1. I don't know if this will work in your
 case, but it's something to try.

 --
 Shaun Ruffell
 Digium, Inc. | Linux Kernel Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at: www.digium.com  www.asterisk.org

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Re: [asterisk-users] Problem installing FXS module in old digium 4 channel tdm card

[C F]

This worked.
Thank you all for your help.

On Wed, Feb 23, 2011 at 1:42 PM, Greg Woods g...@gregandeva.net wrote:
 On Wed, 2011-02-23 at 09:56 -0500, C F wrote:
 This is the closest thing I was able to find in my wctdm.c file:
         if ((blah  0xf) == 2) {
                 /* ProSLIC 3215, not a 3210 */
                 wc-flags[card] |= FLAG_3215;
         }
 If I take out the 2 first lines I get errors when compling.


 Maybe you need to remove the closing brace too?

 --Greg




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Re: [asterisk-users] Problem installing FXS module in old digium 4 channel tdm card

[C F]

How/Where would I do that?

TIA
CF

On Mon, Feb 21, 2011 at 11:23 PM, Shaun Ruffell sruff...@digium.com wrote:
 On 2/21/11 4:46 PM, C F wrote:

 I just installed an FXS module onto a 4 channel tdm thats about 5
 years old and it wont work. Running dmesg I can see the following
 error:

 Zapata Telephony Interface Registered on major 196
 Freshmaker version: 71
 Freshmaker passed register test
 !!! LOOP_CLOSE_TRES  iREG 1C = 1  should be 1000
 !!! RING_TRIP_TRES  iREG 1D = 8000  should be 3600
 !!! COMMON_MIN_TRES  iREG 1E = 0  should be 1000
 !!! COMMON_MAX_TRES  iREG 1F = 0  should be 200
 !!! PWR_ALARM_Q1Q2  iREG 20 = 1480  should be 7C0
 !!! PWR_ALARM_Q3Q4  iREG 21 = 37C0  should be 2600
 !!! PWR_ALARM_Q5Q6  iREG 22 = 3D70  should be 1B80
 !!! LOOP_CLOSURE_FILTER  iREG 23 = 3970  should be 8000
 !!! RING_TRIP_FILTER  iREG 24 = 78E0  should be 320
 !!! TERM_LP_POLE_Q1Q2  iREG 25 = 8B60  should be 8C
 !!! TERM_LP_POLE_Q3Q4  iREG 26 = 6A40  should be 100
 !!! TERM_LP_POLE_Q5Q6  iREG 27 = 8070  should be 10
 !!! CM_BIAS_RINGING  iREG 28 =   should be C00
 !!! DCDC_MIN_V  iREG 29 =   should be C00
 !!! DCDC_XTRA  iREG 2A =   should be 1000
 !!! LOOP_CLOSE_TRES_LOW  iREG 2B =   should be 1000
  ! Init Indirect Registers UNSUCCESSFULLY.
 Indirect Registers failed verification.
 !!! LOOP_CLOSE_TRES  iREG 1C = 1  should be 1000
 !!! RING_TRIP_TRES  iREG 1D = 8000  should be 3600
 !!! COMMON_MIN_TRES  iREG 1E = 0  should be 1000
 !!! COMMON_MAX_TRES  iREG 1F = 0  should be 200
 !!! PWR_ALARM_Q1Q2  iREG 20 = 1480  should be 7C0
 !!! PWR_ALARM_Q3Q4  iREG 21 = 37C0  should be 2600
 !!! PWR_ALARM_Q5Q6  iREG 22 = 3D70  should be 1B80
 !!! LOOP_CLOSURE_FILTER  iREG 23 = 3970  should be 8000
 !!! RING_TRIP_FILTER  iREG 24 = 78E0  should be 320
 !!! TERM_LP_POLE_Q1Q2  iREG 25 = 8B60  should be 8C
 !!! TERM_LP_POLE_Q3Q4  iREG 26 = 6A40  should be 100
 !!! TERM_LP_POLE_Q5Q6  iREG 27 = 8070  should be 10
 !!! CM_BIAS_RINGING  iREG 28 =   should be C00
 !!! DCDC_MIN_V  iREG 29 =   should be C00
 !!! DCDC_XTRA  iREG 2A =   should be 1000
 !!! LOOP_CLOSE_TRES_LOW  iREG 2B =   should be 1000
  ! Init Indirect Registers UNSUCCESSFULLY.
 Indirect Registers failed verification.
 Module 0: FAILED FXS (FCC)
 Module 1: Installed -- AUTO FXO (FCC mode)
 Module 2: Installed -- AUTO FXO (FCC mode)
 Module 3: Installed -- AUTO FXO (FCC mode)
 Found a Wildcard TDM: Wildcard TDM400P REV E/F (3 modules)

 Does this have to do with the fact that the module is way newer than the
 card?


 Not having much direct experience with the wctdm.c driver, that would be my
 guess. You might be able to edit the wctdm_proslic_insane() function to
 force the FLAG_3215 on for the card and see if that gives you a different
 result.

 --
 Shaun Ruffell
 Digium, Inc. | Linux Kernel Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at: www.digium.com  www.asterisk.org

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[asterisk-users] Problem installing FXS module in old digium 4 channel tdm card

[C F]

I just installed an FXS module onto a 4 channel tdm thats about 5
years old and it wont work. Running dmesg I can see the following
error:

Zapata Telephony Interface Registered on major 196
Freshmaker version: 71
Freshmaker passed register test
!!! LOOP_CLOSE_TRES  iREG 1C = 1  should be 1000
!!! RING_TRIP_TRES  iREG 1D = 8000  should be 3600
!!! COMMON_MIN_TRES  iREG 1E = 0  should be 1000
!!! COMMON_MAX_TRES  iREG 1F = 0  should be 200
!!! PWR_ALARM_Q1Q2  iREG 20 = 1480  should be 7C0
!!! PWR_ALARM_Q3Q4  iREG 21 = 37C0  should be 2600
!!! PWR_ALARM_Q5Q6  iREG 22 = 3D70  should be 1B80
!!! LOOP_CLOSURE_FILTER  iREG 23 = 3970  should be 8000
!!! RING_TRIP_FILTER  iREG 24 = 78E0  should be 320
!!! TERM_LP_POLE_Q1Q2  iREG 25 = 8B60  should be 8C
!!! TERM_LP_POLE_Q3Q4  iREG 26 = 6A40  should be 100
!!! TERM_LP_POLE_Q5Q6  iREG 27 = 8070  should be 10
!!! CM_BIAS_RINGING  iREG 28 =   should be C00
!!! DCDC_MIN_V  iREG 29 =   should be C00
!!! DCDC_XTRA  iREG 2A =   should be 1000
!!! LOOP_CLOSE_TRES_LOW  iREG 2B =   should be 1000
 ! Init Indirect Registers UNSUCCESSFULLY.
Indirect Registers failed verification.
!!! LOOP_CLOSE_TRES  iREG 1C = 1  should be 1000
!!! RING_TRIP_TRES  iREG 1D = 8000  should be 3600
!!! COMMON_MIN_TRES  iREG 1E = 0  should be 1000
!!! COMMON_MAX_TRES  iREG 1F = 0  should be 200
!!! PWR_ALARM_Q1Q2  iREG 20 = 1480  should be 7C0
!!! PWR_ALARM_Q3Q4  iREG 21 = 37C0  should be 2600
!!! PWR_ALARM_Q5Q6  iREG 22 = 3D70  should be 1B80
!!! LOOP_CLOSURE_FILTER  iREG 23 = 3970  should be 8000
!!! RING_TRIP_FILTER  iREG 24 = 78E0  should be 320
!!! TERM_LP_POLE_Q1Q2  iREG 25 = 8B60  should be 8C
!!! TERM_LP_POLE_Q3Q4  iREG 26 = 6A40  should be 100
!!! TERM_LP_POLE_Q5Q6  iREG 27 = 8070  should be 10
!!! CM_BIAS_RINGING  iREG 28 =   should be C00
!!! DCDC_MIN_V  iREG 29 =   should be C00
!!! DCDC_XTRA  iREG 2A =   should be 1000
!!! LOOP_CLOSE_TRES_LOW  iREG 2B =   should be 1000
 ! Init Indirect Registers UNSUCCESSFULLY.
Indirect Registers failed verification.
Module 0: FAILED FXS (FCC)
Module 1: Installed -- AUTO FXO (FCC mode)
Module 2: Installed -- AUTO FXO (FCC mode)
Module 3: Installed -- AUTO FXO (FCC mode)
Found a Wildcard TDM: Wildcard TDM400P REV E/F (3 modules)

Does this have to do with the fact that the module is way newer than the card?

TIA
C F

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Re: [asterisk-users] Hide the plain text password

[C F]

On Tue, Feb 15, 2011 at 10:31 AM, Danny Nicholas da...@debsinc.com wrote:
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
 Sent: Tuesday, February 15, 2011 9:29 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Hide the plain text password


 Security through obscurity does not work with open source software.


 What a bold statement, are you telling me it works with closed source
 software? :P


I love this, here you go, security through obscurity at its best:
http://www.feplaw.com/news/lawsuit-filed-against-kaba-ilco20110211.cfm

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Re: [asterisk-users] Hide the plain text password

[C F]

 Security through obscurity does not work with open source software.


What a bold statement, are you telling me it works with closed source
software? :P

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Re: [asterisk-users] Transfer Device Data

[C F]

${BLINDTRANSFER} should hold the device name of the one doing the
blind transfer.


On Sat, Feb 12, 2011 at 6:06 PM, Elliot Murdock murdo...@gmail.com wrote:
 Hello!

 I am trying to find out the device name and/or other identifying data
 to be used in a context when a device transfers the call to new a
 phone number.  From running tests, it looks like the account code
 variable (${CDR(accountcode)}) is set to the account code of the
 device that placed the original call, so if the callee device (not the
 original calling device) is making the transfer to a new number, the
 account code will not be correct, since it will be the account code of
 the calling device, but not the called device.

 How do I find out which device is making the transfer?

 Thanks for any suggestions!
 Elliot

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Re: [asterisk-users] PRI voice optimization

[C F]

On Fri, Feb 4, 2011 at 12:41 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
 Hi All,

 This posting regarding PRI voice optimization, on dahdi 2.1.0.4.

 we have more than 4 machine running on 4 port PRI card with echo
 cancellation hardware based.

 i have enabled echo cancel from chan_dahdi.conf using echocancel=yes, now
 more than 70% of call get good voice
 but some of calls having issue for callquality and other voice related
 issues. now my question is that is there
 any voice related parameter that we need to set for INDIA specific region
 and is ther any voice hardware tester for PRI
 that we can use and tell us our PRI [telco] provider that problem is not
 from our side. let give some idea . below are my configuration as well.


If 70% of calls get good quality then chances are its not your problem
for the 30%. Things to look at (for the 30%):
1. Any specific internal phones that this problem sticks with?
2. Other end a cell phone? or maybe VoIP?
3. Any bluetooth involved? Bluetooth IMHO is a disaster of a
technology when it comes to realtime voice as in phone conversations.
Worse than G.729. It should never be used for a professional business
conversation, wired headsets for cell phones still beat any wireless
solutions. For desk phones proprietary RF is far better than BT.
4. What type of call quality? A. Garbled as in under water (jitter???)
or B. Echo as in hearing yourself back after some ms? or C. Static
Select case
case A your end or far end?
case B your end or far end?
case C probably far end.

Hope this helps

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Re: [asterisk-users] end a call after a specific time period

[C F]

 Channel              Location             State   Application(Data)
 SIP/NTT00-   99449046902115@vicid Down    AppDial((Outgoing Line))
 Local/99449046902115 99449046902115@defau Up      
 Dial(SIP/NTT00/449046902115||o
 Local/99449046902115 8302@default:2       Up      Playback(conf)


 After a few seconds it shows

 0*CLI core show channels
 Channel              Location             State   Application(Data)
 SIP/NTT00-   8302@default:2       Up      Playback(conf)
 1 active channel
 1 active call


 Now the conf file is not that long that it keeps on playing.

 I dont know what else to use to end the call.


Post your dialplan please

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Re: [asterisk-users] Unknow T callerid

[C F]

I guess that was the CallerID transmitted by the calling channel.

On Mon, Jan 24, 2011 at 7:31 PM, Jose Flores Galicia floj...@gmail.com wrote:
 Hi List.

 Have any of you guys ever see an incoming call throught Dahdi channel which
 has an callerid T.

 I know whenever is a private call, it shows  as callerid, but what does it
 mean a T callerid?

 Best Regards
 --
 Jose Flores Galicia

 floj...@gmail.com
 BriefCode  Code Based Training

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Re: [asterisk-users] end a call after a specific time period

[C F]

I believe absolute timeout will do that.
http://www.voip-info.org/wiki/view/Asterisk+func+timeout



On Sun, Jan 23, 2011 at 2:04 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.com wrote:
 Hello all,
 I am trying to end a call after a specific time period for that reason i
 have tried various options like using S, L in the dial command. But in vain.
 Now i am thinking to end the call using the AMI... but i am unable to get
 the current active channel.
 . i.e SIP/NT000 when i ask for getchannel it return some thing like
 this Channel=Local/99449070380109@default-f08b,2
 As i have to use this channel name for hangup... Could some one let me know
 how to get channel name or any other way to end call..
 Thanks in advance

 --
 Best Regards
 Shakeel Abbas


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Re: [asterisk-users] Top Posting

[C F]

On Sun, Jan 16, 2011 at 9:47 PM, James Miller paramedi...@gmail.com wrote:
 When you get over 500 emails a day on your blackberry you have make a 
 decision on what is or is not worth reading at that moment.

 Its not lazy at all its cutting through the fluff and finding the emails 
 worth while.  When inside outlook you don't have the hot key b to scroll to 
 the bottom so again, I'd have to scroll down. Add up the time it takes per 
 email x 500 emails, you loose considerable amount of productivity.

Oh C'mon this is definitely lazy never heard of CTRL+END it works in Outlook


 Top posting has its useful place as well as bottom posting.


 Sent from my Verizon BlackBerry. Always on, Always Connected

 -Original Message-
 From: Fred Posner f...@teamforrest.com
 Sender: asterisk-users-boun...@lists.digium.com
 Date: Sun, 16 Jan 2011 21:43:00
 To: Asterisk Users Mailing List - Non-Commercial 
 Discussionasterisk-users@lists.digium.com
 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
        asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Top Posting

 On Mon, 2011-01-17 at 02:31 +, James Miller wrote:
 I hate to disagree but I find it much, much easier to follow conversations 
 when the newest reply is on top.  I find it too time consuming to scroll 
 through a long message just to find out someone left a three word reply.

 As I am on my blackberry more than I am at a pc, if I don't see the reply as 
 soon as I open the message it gets deleted without being read.  Time is 
 money and I don't have time to scroll through every message.

 I will agree that sometimes it is helpful to make replies at the bottom and 
 I will attempt to keep the peace by posting at the bottom when I can, but 
 top posting is easier and more clean to read than having 100 lines of  and 
 broken lines.

 Warmest regards,
 James


 Sent from my Verizon BlackBerry. Always on, Always Connected

 -Original Message-
 From: Lesly Dorval lador...@yahoo.com
 Sender: asterisk-users-boun...@lists.digium.com
 Date: Mon, 17 Jan 2011 02:14:54
 To: asterisk-users@lists.digium.com
 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
       asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Top Posting

 Shaun Ruffell sruffell at digium.com writes:

 
  Whatever your preferred style, the following post is at least worth
  considering.
 
  http://brooksreview.net/2011/01/interleaved-email/
 
  My belief is that it would be nearly impossible for me to follow a high
  volume list if top posting was the preferred style.  For example, the
  following email from the LKML would need to be more verbose if all the
  participants were top posting, because they would all have to set the
  context for their comments.  Instead, you can follow the chain of
  thought for each of the threads contained in the email.
 
  http://article.gmane.org/gmane.linux.kernel/1087665
 
  Anyway, just something to consider,
  Shaun
 I could never understand the strong objection regarding top-posting until 
 Shaun
 shared these examples - though I had been reading lists for more years than I
 care to admit.  These examples clearly show how snipping and bottom posting
 translate to susccint and clear contextual communication. From now I will
 evangelize snipping and bottom posting.



 I cannot imagine considering scrolling to the end of an email time
 consuming. Very sad. If you find it too difficult on your blackberry to
 press the B key (to jump to the bottom of the message) then I am
 uncertain how you have enough time to even read this email.

 I'm all for good arguments. That time consuming one is just lazy.

 I personally find top posting annoying and only serving to an immediate
 conversation. Particularly useless if referencing the message later.

 --

 With best regards,

 ---fred
 http://qxork.com


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Re: [asterisk-users] Benefit of PRI vs SIP trunk calls

[C F]

PRICAUSE will give you lots of info on why a call was hungup on. Not
sure if SIP will give you the same.

On Thu, Jan 6, 2011 at 9:06 AM, Jim Dickenson dicken...@cfmc.com wrote:
 Does Asterisk, currently using version 1.4, get any more information about 
 the result of an outbound call made over a PRI line compared to a call via a 
 SIP trunk?

 As an example, in a PRI call there is this message that shows up on the 
 console:

 [2011-01-05 14:59:02]     -- Channel 23 detected a CED tone from the network.

 for a call to a fax machine. Does asterisk set anything that a dialplan can 
 access that can know the call was to a fax machine?

 If a call is placed to a number that is disconnected so a special information 
 tone is played can either a PRI call or a SIP call know this without 
 analyzing the audio stream?

 Are there reasons to prefer the use of PRI over SIP or SIP over PRI?

 I would like people's opinions as to if one form is better than the other in 
 any meaningful way.

 Thanks for you feed-back.
 --
 Jim Dickenson
 mailto:dicken...@cfmc.com

 CfMC
 http://www.cfmc.com/




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Re: [asterisk-users] (no subject)

[C F]

Anyone going to remove this spammer/scammer?

2010/12/19 Dmitry Kupchinetsky dkupchinet...@hotmail.com:
 http://www.barenakedbabies.com/shop/images/images.html

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Re: [asterisk-users] Voicemail Forwarding

[C F]

Is that user trying to forward to xxx in the same context?

On Fri, Dec 17, 2010 at 5:57 AM, --[ UxBoD ]-- ux...@splatnix.net wrote:
 Experiencing a problem when users attempt to forward a voicemail from within 
 VoiceMailMain(Option 8) I see the console message:

 Couldn't not find mailbox XXX in context default

 As why are running in a multi-tenant environment voicemail.conf has been 
 separated into individual contexts.  The users retrieve their email by 
 dialing an extension which calls VoiceMailMail(x...@vmcontext) so how do I 
 instruct Asterisk to use that context when forwarding voicemails ?
 --
 Thanks, Phil

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Re: [asterisk-users] How to find , internal, external inbound or outbound

[C F]

This list is being attacked by some Khaled I guess. How can we stop him?

On Fri, Dec 17, 2010 at 9:34 AM, Khaled W. Chehab kche...@xplorium.com wrote:
 Hi,

 My system been attacked from someone I guess, kindly check the link below

 How can I stop the ircd attack

 http://pastebin.com/tbjh5qzP



 regards





 
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Re: [asterisk-users] Version compatibility question...

[C F]

Thanks again Kevin
Have a wonderful day :)

On Tue, Dec 7, 2010 at 8:01 AM, Kevin P. Fleming kpflem...@digium.com wrote:
 On 12/06/2010 08:12 PM, C F wrote:
 Thanks Kevin.
 Upto which version fo Dahdi works with 1.4.x?

 If I understand your question properly, all versions of DAHDI are
 compatible with 1.4.x. All versions of DAHDI are backward compatible.

 --
 Kevin P. Fleming
 Digium, Inc. | Director of Software Technologies
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 skype: kpfleming | jabber: kflem...@digium.com
 Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] Version compatibility question...

[C F]

Thanks Kevin.
Upto which version fo Dahdi works with 1.4.x?


On Mon, Dec 6, 2010 at 6:25 PM, Kevin P. Fleming kpflem...@digium.com wrote:
 On 12/05/2010 08:25 AM, C F wrote:
 Is there any version matching doc? since it was changed to Dahdi I
 don't really know which version works with which.

 Asterisk 1.4.21 and lower can only use Zaptel. Asterisk 1.4.22 through
 1.4.x can use either Zaptel or DAHDI. Asterisk 1.6.x, Asterisk 1.8.x and
 all future versions only support DAHDI.

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 Kevin P. Fleming
 Digium, Inc. | Director of Software Technologies
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 skype: kpfleming | jabber: kflem...@digium.com
 Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] Version compatibility question...

[C F]

Is there any version matching doc? since it was changed to Dahdi I
don't really know which version works with which.

On Sun, Dec 5, 2010 at 12:35 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
 On Thu, Dec 02, 2010 at 09:09:25PM -0300, equis software wrote:
 Hi, Could I install Asterisk 1.4.19,  Dahdi 2.4.0 and libpri 1.4.3 ??

 No. Asterisk  1.4.22 cannot use DAHDI.

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               Tzafrir Cohen
 icq#16849755              jabber:tzafrir.co...@xorcom.com
 +972-50-7952406           mailto:tzafrir.co...@xorcom.com
 http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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