Re: [asterisk-users] fax / t38 gateway

[Kristian Kielhofner]

On 10/31/08, Jonn R Taylor <[EMAIL PROTECTED]> wrote:
> Here is the QOS script that I use on my bridge.
>
>  http://www.taylortelephone.com/asterisk/astshape

  You should upgrade to the newer astshape script.  It classifies
traffic using iptables, which is much more flexible.  It also has beta
support for the HFSC qdisc:

http://astlinux.svn.sourceforge.net/viewvc/astlinux/trunk/package/iproute2/astshape

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com

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Re: [asterisk-users] Enter Value and continue dialplan

[Stuart Sheldon]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Take a look at Read()

Stu


David Klaverstyn wrote:
> Hi,
> 
>  
> 
> What function or application do I use to get people to type digits into
> the phone and store the value into a variable?  The application
> WaitExten is not what I want that will jump to the new extension.
> 
>  
> 
> I want users to enter a number into the phone and store it as a variable
> so I can use it later in the dial plan.
> 
>  
> 
> 
> 
> 
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> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.9 (GNU/Linux)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iEYEARECAAYFAkkKk5YACgkQK69Y+xPZrWYUOgCfR59WPrAzCUX321R/tXgMYHUO
BOEAni3KIHrJl7pjUEtrertR9J4gFCSQ
=e2MZ
-END PGP SIGNATURE-

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[asterisk-users] Enter Value and continue dialplan

[David Klaverstyn]

Hi,

 

What function or application do I use to get people to type digits into the
phone and store the value into a variable?  The application WaitExten is not
what I want that will jump to the new extension.

 

I want users to enter a number into the phone and store it as a variable so
I can use it later in the dial plan.

 

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Re: [asterisk-users] fax / t38 gateway

[Jonn R Taylor]

Here is the QOS script that I use on my bridge.

http://www.taylortelephone.com/asterisk/astshape

I have also had a very high success rate with 
Fax-->ATA-->SIP-->Asterisk-->SIP-->PSTN and the other way. The fax is a Brother 
MFC-440CN.

I have posted most of my hylafax iaxmodem configs and other asterisk setup 
scripts. All are welcome.

Steve, you have done a wonder full job on your spandsp library! THANK YOU
I have the ability to do a lot of pure VOIP testing if you need it.

Jonn

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood
Sent: Thursday, October 30, 2008 6:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] fax / t38 gateway

Jonn R Taylor wrote:
> I have been able to repeat the results at other locations. The location that 
> has 26 pages is a linksys PAP2T our accounting person uses remotely to fax 
> stuff to the office. The ATA is behind a DIL-625 router with QOS on a DSL 
> line. 
>
> I can send faxes from my test sever at home that is using an IAX trunk to our 
> office, same ATA. 
> FAX--->ATA--->SIP--->Asterisk/gateway/QOS_shaper--->cablemodem--->internet--->cablemodem_office--->QOS_bridge--->asterisk/iaxmodem-hylafax
>  So, if you look at the protocols that are used it is 
> SIP--->IAX--->SIP--->PSTN or SIP--->IAX--->IAXmodem 
>
> This same connection is currently handling 4000 emails a day, webmail, POP, 
> IMAP, MAPI, VPN traffic, web traffic, and normal web surfing with 
> downloading. One test that I did with the download is started a 650MB iso 
> download at about 900kB. Now at the same time started to send and receive 
> faxes at the same time and worked.
>
> Jonn
>   
That signal chain is probably part of the reason you get away with this 
so often. iaxmodem does not have the real time constraints of a real FAX 
machine. Its still real time, but not as tightly constrained to a smooth 
flow of data. If packets are delayed, and jitter is high, iaxmodem can 
be very tolerant, as long as the packet loss is really low. Your QoS 
should groom things in the other direction, and ensure a reasonably 
smooth outward flow of packets.

Steve


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Re: [asterisk-users] autodialed call forwarding via meetme or queue (was predictive dialer)

[Roi Stork]

Additional question: are there instances when the incoming call waiting in
the queue is dropped when connected to a waiting agent/local extension?

By the way, incoming call channel is: Local/[EMAIL PROTECTED]

created via Originate

On Sun, Oct 26, 2008 at 10:19 PM, Roi Stork <[EMAIL PROTECTED]> wrote:

> Also posting this question to people working on manager interface and
> dialers.
>
> I have a simple auto dialing script (using Originate) that forwards all
> incoming calls to a queue full of waiting agents instead of a meetme
> conference room. I use queues rather than meetme so I can leave the
> automatic call distribution to the queue itself.
>
> The problem is when the calls reach the agents, some of the agents notice
> that the other line is silent. The queue is already set up to hold an
> infinite number of calls (meaning: maxlen=0/no limit), and the agents are
> already answering the calls immediately/after one ring, but the problem
> still shows up.
>
> Is forwarding to a meetme conference room faster than through a queue?
>
> On Thu, Oct 16, 2008 at 11:25 PM, Steve Totaro <
> [EMAIL PROTECTED]> wrote:
>
>> If you can figure out how to generate .call files from your DB
>> entries, you have it made.
>>
>> Vicidial needs alot of work as far as I am concerned, for free it is
>> OK I guess.  I think using meetme conference rooms for everything is a
>> kludgy hack, and the UI is less than nice (if you are into UIs).
>>
>> I suggest you continue on your own custom development if you have the
>> time.  Check out Aheeva for inspiration.
>>
>> Thanks,
>> Steve Totaro
>>
>> On Fri, Oct 17, 2008 at 1:31 AM, ram <[EMAIL PROTECTED]> wrote:
>> > look at Vicidial
>> >
>> > ram
>> >
>> > On Thu, Oct 16, 2008 at 4:46 PM, yavuz yildirim <[EMAIL PROTECTED]>
>> wrote:
>> >>
>> >> hi everybody
>> >>
>> >> This is Yavuz YILDIRIM
>> >>
>> >> I am software developer.I have a some problems in asterisk.
>> >> I am using mysql db. Realtime using asterisk modules. On db i am using
>> >> calling hundred fields for use dial.
>> >> But i don't know how i can automaticly dial this fields on records
>> >> numbers. Who can help me asterisk api and others.
>> >>
>> >> Thank you
>> >>
>> >>
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>>
>>
>>
>> --
>> Thanks,
>> Steve Totaro
>> +18887771888 (Toll Free)
>> +12409381212 (Cell)
>> +12024369784 (Skype)
>>
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>
>
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[asterisk-users] Asterisk with SC440 Dell(Big Problem)

[Edwin Quijada]

I have a Dell SC440 with Centos and Asterisk 1.4.21 and a card openvox D110PG, 
T1, when a person calling from the PTSN will listen to them but then begins to 
distort the voice I heard that name. I probe the card in another computer and 
it works perfectly. Anyone has any idea or help. I'm going crazy with this 
problem. Install Debian on this server and the same thing happened to me.
I bought this server and now it doesnt work with asterisk.
I will appreciate if somebody has any cluee or idea about this.

If anybody has this server i'd like to know everything about your config. 
TIA

*---*
*-Edwin Quijada
*-Developer DataBase

_
Get Windows Live and get whatever you need, wherever you are.  Start here.
http://www.windowslive.com/default.html?ocid=TXT_TAGLM_WL_Home_082008
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Re: [asterisk-users] Decent Voip Phones for enterprise

[Paul Hales]

I know a business that tried those phones, and removed them.

They found that Polycom phones were 'more' perfect.

PaulH


Bruno Castelo Branco wrote:
> hi
> O use around 500 atcom530, they are work perfect
> www.atcom.com.cn
>
> Gordon Henderson wrote:
>> On Wed, 29 Oct 2008, Kev Szaszvari wrote:
>>
>>   
>>> Hi there
>>>
>>> Our company is using the Linksys SPA-942 Phones, and they are pretty 
>>> useless.
>>> They dont have any central management or provisioning, as well as a pretty 
>>> bad interface.
>>>
>>> Can anyone reccomend any voip phones( Cisco, Polycom, SNOM ) that have
>>>
>>> * Central Management for all the phones (We dont mind if we have to buy 
>>> the software to manage them)
>>> 
>>
>> I always wondered about this - my target is the SME - say 4-150 seats - 
>> people don't move desks, change office that often, staff "churn" is 
>> typically low, so I program the phones once then leave them there. If you 
>> move desk you take your phone with you. If you leave then the phone can be 
>> renamed via it's web interface relatively easily.
>>
>> Maybe I'm just dealing with simple (dumb?) offices, but I'm curious as to 
>> what people do with the phones that require this sort of central 
>> management. (And regular phone updating)
>>
>>   
>>> * Programable shortcut buttons, So i can program in on certian phones 
>>> quick dials to queues.
>>> 
>>
>> How about implementing this in the PBX.
>>
>>   
>>> * Optional but bonus, The ability to have a shared address book accross 
>>> the phones.
>>> 
>>
>> Same here.
>>
>> So some phones do have nice programmable buttons - and that's good, but in 
>> my PBX I have the space for about 600 speed-dials (3-digit extensions) 
>> which are web managed by the admin, and 30 personal ones settable on the 
>> phone *00 through *29 ... (I know this sometimes might clash with a phones 
>> own 'star' codes though)
>>
>> But maybe this is just me ... When I started playing with asterisk I 
>> bought a small number of different phones to get a feel for them and was 
>> frustrated by a lack of common functions across them, so put all features 
>> back into the PBX - things like diverts, follow-me, voicemail and so on 
>> are all handled by my asterisk system rather than relying on a particular
>> SIP phone to handle it...
>>
>> However if you want to know what phones I use, it's mostly Grandstream for 
>> now. I provision them using gsutil, and when customers want something a 
>> bit more posh, it's Snoms.
>>
>> Gordon
>>
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> 
>
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Re: [asterisk-users] network design philosophy and practice

[Paul Hales]

Separate cabling is also useful if the phone system is being deployed by
a separate company - it avoids the 'your computer network is generating
rubbish traffic' arguments. (been there before, sadly)

PaulH


Andrew Latham wrote:
> Alex
>
> I see a fair bit of separate physical networks because of different
> management of phones vs IT.  In the old businesses Facilities handles
> the communications and IT is playing catchup all the time
>
> So in these businesses where the IT side is swapping switches on a
> weekly basis it is safer to have a separate physical network.
>
>
> Andrew
>
>
> On Wed, Oct 29, 2008 at 11:30 AM, Alex Balashov
> <[EMAIL PROTECTED]> wrote:
>   
>> I'm pretty sure they meant two logical networks.  At least, I hope they did.
>>
>> David Gibbons wrote:
>>
>> 
>>> Two separate networks? Did I miss something? I feel like I'm taking crazy 
>>> pills! Two separate physical networks means twice the hassle, twice the 
>>> maintenance, twice the cost, twice the headache. Not to mention the fact 
>>> that the whole idea of VOIP is to simplify IT and focus on converging data 
>>> and voice networks.
>>>
>>> This is what VLANs and QOS do best. I dare say it's what they were designed 
>>> foe. I can't think of any reason that I would ever recommend two ports per 
>>> desk to support telephony -- ever. It's ludicrous to think that two ports 
>>> will be better than one if we're setting up our VLANs and QOS properly. A 
>>> phone takes very, very little bandwidth away from the desktop and a decent 
>>> one will support tagging its frames for the alternate voice VLAN.
>>>
>>> --snip--
>>> In almost all cases it is much better to have two seperate networks.
>>> This may be impractical in some smaller installs, but in any office
>>> setting we always do this. The only reason I can think of not to is to
>>> eliminate the cost of the second cable.
>>> --snip--
>>>
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>>>   
>> --
>> Alex Balashov
>> Evariste Systems
>> Web: http://www.evaristesys.com/
>> Tel: (+1) (678) 954-0670
>> Direct : (+1) (678) 954-0671
>> Mobile : (+1) (706) 338-8599
>>
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>> 
>
>
>
>   


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Re: [asterisk-users] fax / t38 gateway

[Steve Underwood]

Jonn R Taylor wrote:
> I have been able to repeat the results at other locations. The location that 
> has 26 pages is a linksys PAP2T our accounting person uses remotely to fax 
> stuff to the office. The ATA is behind a DIL-625 router with QOS on a DSL 
> line. 
>
> I can send faxes from my test sever at home that is using an IAX trunk to our 
> office, same ATA. 
> FAX--->ATA--->SIP--->Asterisk/gateway/QOS_shaper--->cablemodem--->internet--->cablemodem_office--->QOS_bridge--->asterisk/iaxmodem-hylafax
>  So, if you look at the protocols that are used it is 
> SIP--->IAX--->SIP--->PSTN or SIP--->IAX--->IAXmodem 
>
> This same connection is currently handling 4000 emails a day, webmail, POP, 
> IMAP, MAPI, VPN traffic, web traffic, and normal web surfing with 
> downloading. One test that I did with the download is started a 650MB iso 
> download at about 900kB. Now at the same time started to send and receive 
> faxes at the same time and worked.
>
> Jonn
>   
That signal chain is probably part of the reason you get away with this 
so often. iaxmodem does not have the real time constraints of a real FAX 
machine. Its still real time, but not as tightly constrained to a smooth 
flow of data. If packets are delayed, and jitter is high, iaxmodem can 
be very tolerant, as long as the packet loss is really low. Your QoS 
should groom things in the other direction, and ensure a reasonably 
smooth outward flow of packets.

Steve


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[asterisk-users] 1.4.22 vs 1.4.21.2 - IAX2 regression ?

[Ex Vito]

Hi list,

I just experienced an odd behaviour in 1.4.22 vs 1.4.21.2.
To cut a long story short, IAX2 is not tx-ing hangup...

Scenario is composed of two asterisk systems A and B.
A receives calls from IAX users X, Y, Z, etc, does some
validation and forwards them to B, also over IAX.

When B hangs up, it transmits IAX hangup which A receives
who, in turn, does not transmit the IAX hangup to its user
X, Y or Z. So X, Y or Z still think the call is up...

All of this is verified with iax debug... A receives the hangup but
never hangs up the other side if running 1.4.22. Everything is ok
if running 1.4.21.2.

Could this be something we're doing wrong ? What steps would
you suggest for further diagnostic?

Thanks in advance for any feedback.



System A runs 1.4.22 / 1.4.21.2

System A iax.conf
[userX]
type=user
transfer=no
host=dynamic
secret=
context=the-context
disallow=all
allow=alaw
allow=ulaw

[systemB]
type=peer
qualify=200
transfer=no
host=
disallow=all
allow=gsm

System A extensions.conf:
[the-context]
exten => _.,1,Wait(1)
exten => _.,n,Set(CALL_UUID=${EXTEN})
exten => _.,n,Set(RESULT_STRING="${ODBC_CALL_DATA_4_UUID(${CALL_UUID})}")
exten => _.,n,Set(ARRAY(NAME,ACCT,IAXUSER,NUM)="${RESULT_STRING}")
exten => _.,n,Set(DONT_CARE="${ODBC_REMOVE_CALL_4_UUID(${CALL_UUID})}")
exten => _.,n,Set(CALLERID(name)=${NAME})
exten => _.,n,Set(CDR(accountcode)=${ACCT})
exten => _.,n,Dial(IAX2/[EMAIL PROTECTED]/${NUM})
exten => _.,n,Hangup()

(note: behaviour is also failing in 1.4.22 if, instead of Dialing
system B, we just wait+hangup directly here!)



System B runs asterisk 1.2.30.1
System B iax.conf:
[one-systemA-user]
type=user
context=one-context
notransfer=yes
disallow=all
allow=gsm

System B extensions.conf:
[one-context]
exten => _N,1,Dial(.../${EXTEN})
exten => _N,n,Hangup()



--
 exvito

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Re: [asterisk-users] Message 245058

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Re: [asterisk-users] CHANUNAVAIL with a TDM800 card

[hin lee]

Thanks for the asterisk restart command.  That saved me a few minutes during 
each test.  As for the genzaptelconf command, it creates zaptel.conf and 
zapata-auto.conf but not the zaptel-channels.conf.  Zaptel-channels.conf is 
blank and doesn't work until I manually add a channel to it.  Thanks for all 
your responses.

[EMAIL PROTECTED]:/ $ genzaptelconf



STOPPING ASTERISK
Asterisk Stopped

STOPPING FOP SERVER
FOP Server Stopped
Generating  '/etc/zaptel.conf'
Generating  '/etc/asterisk/zapata-auto.conf'
Unloading zaptel hardware drivers:.
Loading zaptel framework:  [  OK  ]
Waiting for zap to come online...OK
Loading zaptel hardware modules: tor2.
 wct4xxp.
 wcte12xp.
 wct1xxp.
 wcte11xp.
 wctdm24xxp.
 wcfxo.
 wctdm.
 wcusb.
 xpp_usb.
Running ztcfg: [  OK  ]


SETTING FILE PERMISSIONS
Permissions OK

STARTING ASTERISK
Asterisk Started

STARTING FOP SERVER
FOP Server Started
   Chan Extension  Context Language   MOH Interpret
 pseudofrom-zaptel en default




--- On Thu, 10/30/08, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:

> From: Tzafrir Cohen <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] CHANUNAVAIL with a TDM800 card
> To: asterisk-users@lists.digium.com
> Date: Thursday, October 30, 2008, 1:06 PM
> On Thu, Oct 30, 2008 at 12:56:06PM -0700, hin lee wrote:
> > Tzafrir,
> > 
> > You are correct!  I didn't have to commented out
> the unused FXO ports. So to revise my earlier email, I have
> to do the following:
> > 
> > 1) Run genzaptelconf
> > 
> > 2) Run "cat /proc/zaptel/*" to find the
> channel my line is connected to.
> > 
> > 3) Add my channel to
> /etc/asterisk/zapata-channels.conf
> 
> Why is that? genzaptelconf should generate that file.
> 
> > 
> >  ie. channel => 1
> > 
> >  I'm not sure why I have to do this manually.  My
> zapata-channels.conf file is blank and doesn't work
> until I put the "channel => X" to it.
> > 
> > 4) Of course, reboot the server.
> 
> Why is that? 'dahdi restart' should do. Or in the
> worst case, restart
> asterisk.
> 
> > 
> > 
> > 
> > 
> > --- On Thu, 10/30/08, Tzafrir Cohen
> <[EMAIL PROTECTED]> wrote:
> > 
> > > From: Tzafrir Cohen
> <[EMAIL PROTECTED]>
> > > Subject: Re: [asterisk-users] CHANUNAVAIL with a
> TDM800 card
> > > To: asterisk-users@lists.digium.com
> > > Date: Thursday, October 30, 2008, 11:20 AM
> > > On Thu, Oct 30, 2008 at 11:03:03AM -0700, hin lee
> wrote:
> > > > I got this working.  For what it's
> worth,
> > > here's what the issue.  
> > > > 
> > > > The channel wasn't getting created under
> FreePBX
> > > via script.  Here's what I needed to do:
> > > > 
> > > > 1) Run genzaptelconf  to generate the zaptel
> configs
> > > 
> > > This generates you /etc/zaptel.conf and
> > > /etc/asterisk/zapata-channels.conf (or
> > > /etc/asterisk/zapata-auto.conf ,
> > > in the modified versions by some distributions).
> > > 
> > > > 
> > > > 2) find the channel the port(s) is on.
> > > > 
> > > > cat /proc/zaptel/*
> > > > 
> > > > 3) comment out the unused ports in
> /etc/zaptel.conf
> > > based on step 2 result.
> > > 
> > > Why?
> > > 
> > > > 
> > > > 4) put in the available channel in
> > > /etc/asterisk/zapata-channels.conf
> > > > 
> > > > ie. channel => 1
> > > 
> > > So I gather it has generated for you
> zapata-auto.conf but
> > > zapata.conf
> > > #include-s zapata-channels.conf . You're
> confused (or
> > > someone did some
> > > bad integration work).
> > > 
> > > > 
> > > > 5) comment out the unused channels in
> > > /etc/asterisk/zapata-auto.conf
> > > > 
> > > > 
> > > >
> > >
> http://www.freepbx.org/support/documentation/administration-guide/interfacing-to-a-pstn
> > > 
> > > This thing should be removed. It is aufully
> confusing and
> > > completely
> > > outdated. I think it is slightly worse than no
> information
> > > at all, as
> > > the defaults of freepbx would have worked for you
> if you
> > > just ran
> > > genzaptelconf .
> > > 
> > > -- 
> > >Tzafrir Cohen
> > > icq#16849755 
> jabber:[EMAIL PROTECTED]
> > > +972-50-7952406  
> mailto:[EMAIL PROTECTED]
> > > http://www.xorcom.com 
> iax:[EMAIL PROTECTED]/tzafrir
> > > 
> > > ___
> > > -- Bandwidth and Colocation Provided by
> > > http://www.api-digital.com --
> > > 
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >   
> http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> > 
> >   
> > 
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> >   
> http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> -- 
>Tzafrir Cohen
> icq#16849755  jabber:[EMAIL PROTECTED]
> +972-50-7952406   mailto:[EMAIL PROTECTED]
> http://www.xorc

Re: [asterisk-users] CHANUNAVAIL with a TDM800 card

[Tzafrir Cohen]

On Thu, Oct 30, 2008 at 12:56:06PM -0700, hin lee wrote:
> Tzafrir,
> 
> You are correct!  I didn't have to commented out the unused FXO ports. So to 
> revise my earlier email, I have to do the following:
> 
> 1) Run genzaptelconf
> 
> 2) Run "cat /proc/zaptel/*" to find the channel my line is connected to.
> 
> 3) Add my channel to /etc/asterisk/zapata-channels.conf

Why is that? genzaptelconf should generate that file.

> 
>  ie. channel => 1
> 
>  I'm not sure why I have to do this manually.  My zapata-channels.conf file 
> is blank and doesn't work until I put the "channel => X" to it.
> 
> 4) Of course, reboot the server.

Why is that? 'dahdi restart' should do. Or in the worst case, restart
asterisk.

> 
> 
> 
> 
> --- On Thu, 10/30/08, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> 
> > From: Tzafrir Cohen <[EMAIL PROTECTED]>
> > Subject: Re: [asterisk-users] CHANUNAVAIL with a TDM800 card
> > To: asterisk-users@lists.digium.com
> > Date: Thursday, October 30, 2008, 11:20 AM
> > On Thu, Oct 30, 2008 at 11:03:03AM -0700, hin lee wrote:
> > > I got this working.  For what it's worth,
> > here's what the issue.  
> > > 
> > > The channel wasn't getting created under FreePBX
> > via script.  Here's what I needed to do:
> > > 
> > > 1) Run genzaptelconf  to generate the zaptel configs
> > 
> > This generates you /etc/zaptel.conf and
> > /etc/asterisk/zapata-channels.conf (or
> > /etc/asterisk/zapata-auto.conf ,
> > in the modified versions by some distributions).
> > 
> > > 
> > > 2) find the channel the port(s) is on.
> > > 
> > >   cat /proc/zaptel/*
> > >   
> > > 3) comment out the unused ports in /etc/zaptel.conf
> > based on step 2 result.
> > 
> > Why?
> > 
> > > 
> > > 4) put in the available channel in
> > /etc/asterisk/zapata-channels.conf
> > > 
> > > ie. channel => 1
> > 
> > So I gather it has generated for you zapata-auto.conf but
> > zapata.conf
> > #include-s zapata-channels.conf . You're confused (or
> > someone did some
> > bad integration work).
> > 
> > > 
> > > 5) comment out the unused channels in
> > /etc/asterisk/zapata-auto.conf
> > > 
> > > 
> > >
> > http://www.freepbx.org/support/documentation/administration-guide/interfacing-to-a-pstn
> > 
> > This thing should be removed. It is aufully confusing and
> > completely
> > outdated. I think it is slightly worse than no information
> > at all, as
> > the defaults of freepbx would have worked for you if you
> > just ran
> > genzaptelconf .
> > 
> > -- 
> >Tzafrir Cohen
> > icq#16849755  jabber:[EMAIL PROTECTED]
> > +972-50-7952406   mailto:[EMAIL PROTECTED]
> > http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
> > 
> > ___
> > -- Bandwidth and Colocation Provided by
> > http://www.api-digital.com --
> > 
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
>   
> 
> ___
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> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
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-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] CHANUNAVAIL with a TDM800 card

[hin lee]

Tzafrir,

You are correct!  I didn't have to commented out the unused FXO ports. So to 
revise my earlier email, I have to do the following:

1) Run genzaptelconf

2) Run "cat /proc/zaptel/*" to find the channel my line is connected to.

3) Add my channel to /etc/asterisk/zapata-channels.conf

 ie. channel => 1

 I'm not sure why I have to do this manually.  My zapata-channels.conf file is 
blank and doesn't work until I put the "channel => X" to it.

4) Of course, reboot the server.




--- On Thu, 10/30/08, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:

> From: Tzafrir Cohen <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] CHANUNAVAIL with a TDM800 card
> To: asterisk-users@lists.digium.com
> Date: Thursday, October 30, 2008, 11:20 AM
> On Thu, Oct 30, 2008 at 11:03:03AM -0700, hin lee wrote:
> > I got this working.  For what it's worth,
> here's what the issue.  
> > 
> > The channel wasn't getting created under FreePBX
> via script.  Here's what I needed to do:
> > 
> > 1) Run genzaptelconf  to generate the zaptel configs
> 
> This generates you /etc/zaptel.conf and
> /etc/asterisk/zapata-channels.conf (or
> /etc/asterisk/zapata-auto.conf ,
> in the modified versions by some distributions).
> 
> > 
> > 2) find the channel the port(s) is on.
> > 
> > cat /proc/zaptel/*
> > 
> > 3) comment out the unused ports in /etc/zaptel.conf
> based on step 2 result.
> 
> Why?
> 
> > 
> > 4) put in the available channel in
> /etc/asterisk/zapata-channels.conf
> > 
> > ie. channel => 1
> 
> So I gather it has generated for you zapata-auto.conf but
> zapata.conf
> #include-s zapata-channels.conf . You're confused (or
> someone did some
> bad integration work).
> 
> > 
> > 5) comment out the unused channels in
> /etc/asterisk/zapata-auto.conf
> > 
> > 
> >
> http://www.freepbx.org/support/documentation/administration-guide/interfacing-to-a-pstn
> 
> This thing should be removed. It is aufully confusing and
> completely
> outdated. I think it is slightly worse than no information
> at all, as
> the defaults of freepbx would have worked for you if you
> just ran
> genzaptelconf .
> 
> -- 
>Tzafrir Cohen
> icq#16849755  jabber:[EMAIL PROTECTED]
> +972-50-7952406   mailto:[EMAIL PROTECTED]
> http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
> 
> ___
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> http://www.api-digital.com --
> 
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


  

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Re: [asterisk-users] ISDN - BRI

[Tzafrir Cohen]

On Thu, Oct 30, 2008 at 12:44:40PM -0600, Wilton Helm wrote:
> Subsequent to some previous E-Mails, I've been trying to dig into the ISDN - 
> BRI situation a bit more.  I have determined that I have a HFC card with 
> Winbond chip, but I'm not sure what combination of drivers is best or usable.
> 
> zaphfc is out because it only supports the cologne chip.

The HFC chip is "the cologne chip".

The HFC-S chip is used in various common single-port ISDN cards.

> 
> misdn is a possibility.  I haven't determined if it supports the card 
> natively, or needs a card specific driver under it.
> 
> capi is a possibility, but again, I don't know what driver, if any needs to 
> be under it.
> 
> capi can support misdn under it, but I don't know if this is an advantage or 
> not, and again whether a card driver needs to be under misdn
> 
> libpri 1.4.4 is supposed to work, provided you unpatch the bad patch in the 
> source and compile it--again, I'm not sure what driver, if any needs to be 
> under it.

This is if you have zaphfc.

> 
> F9 detected the card and loaded some sort of driver support for it, but I 
> don't know if that covers the lower levels appropriately.
> 
> Can anyone provide more specific information or suggest which of these 
> approaches is most likely to work, most likely to be stable, or supported for 
> the future, or support the most features?  It would appear there are at least 
> four possible ways that might work, but I can't determine which is best.
> 
> Thanks,
> Wilton

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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[asterisk-users] ISDN - BRI

[Wilton Helm]

Subsequent to some previous E-Mails, I've been trying to dig into the ISDN - 
BRI situation a bit more.  I have determined that I have a HFC card with 
Winbond chip, but I'm not sure what combination of drivers is best or usable.

zaphfc is out because it only supports the cologne chip.

misdn is a possibility.  I haven't determined if it supports the card natively, 
or needs a card specific driver under it.

capi is a possibility, but again, I don't know what driver, if any needs to be 
under it.

capi can support misdn under it, but I don't know if this is an advantage or 
not, and again whether a card driver needs to be under misdn

libpri 1.4.4 is supposed to work, provided you unpatch the bad patch in the 
source and compile it--again, I'm not sure what driver, if any needs to be 
under it.

F9 detected the card and loaded some sort of driver support for it, but I don't 
know if that covers the lower levels appropriately.

Can anyone provide more specific information or suggest which of these 
approaches is most likely to work, most likely to be stable, or supported for 
the future, or support the most features?  It would appear there are at least 
four possible ways that might work, but I can't determine which is best.

Thanks,
Wilton
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[asterisk-users] General development funding: discussion and survey

[John Todd]

[sending to -users and -biz in a slightly different format to broaden  
participation]

Summary:
   Would you help fund different Open-Source Asterisk enhancements,  
bugfixes, or documentation if there was a way to collectively  
contribute money towards the effort without a profit margin  
incorporated into the price?  If so, jump to the bottom of this  
message and fill out the form on the URL provided.   There is no  
obligation to anything by filling out the form - think of this as a  
"market survey".


The Long Version:

Everyone on this list is presumably an Asterisk user, advocate, or is  
in some way benefitting from the project.  Your ideas and survey  
participation would be welcome on the topic below.

Many coders love coding for Asterisk but often can't find the time to  
do it for free when faced with things like buying food, paying  
mortgages, and keeping current with their insurance - this is totally  
understandable.  Many coders have and continue to contribute things to  
Asterisk at no cost, but these patches are typically their own  
"itches", where they have solved a particular problem of their own.   
Rarely do people pick up problems that are not related to anything  
they're doing, or pick up unrelated problems that are so large that it  
would involve 100% of their time for any significant period.  Some  
people ("Bless your heart!" as they say here in Huntsville) work on  
bugs and enhancements that don't directly benefit them at all - these  
are the most valuable contributors we have - you know who you are.
Most of the time, though, there is a directly relevant reason why  
people work on code and often that means more obscure bugs or feature  
implementations languish, though still worthwhile if someone were to  
complete them.

On the other side of the scale there are many people or companies who  
perhaps would like to contribute to paying for various features in  
Asterisk that would be described as "large enhancements" or even minor  
bugs and annoyances, but do not have sufficient funds to pay for an  
entire project themselves.  There are perhaps also many people who  
would like to help out Asterisk in a way that allows them to  
contribute funding towards the project, but they're uncomfortable  
sending money to a corporation and hoping that it gets eventually  
applied to OSS Asterisk (and I'm not only talking about Digium in this  
case.)   There are coders available for a fee (perhaps much less than  
market rate, perhaps not - we'll just say "non-zero cost") who could  
do this work and would love to do it if they could justify the time  
spent.   Open-Source Software doesn't always imply that the code is  
"unpaid work", and Digium's contributions towards Asterisk are a case  
for the benefits of having an income stream and payment system  
(salaries) that supplements OSS development.

So there is a disconnect between two groups of willing consumers and  
willing producers - how do we bridge it?  The answer some have come up  
with is "Let's create an Asterisk fund and collect money and disperse  
money to pay for work by community members!"  This is a great concept,  
but the devil is in the details, and I've found that when money is  
involved, the detail devil is much larger and angrier than usual.


The problems with this idea have continually been:

  - Escrow of capital.  It is not feasible to trust that donors will  
be good on their contribution post-release.  This may be because it  
takes a while for the code and economic situations change, it may be  
that internal paperwork processes take forever to get done (Hi, Raj,  
sorry about that delay from Tello!), or it may just be that a large  
portion of funders are flaky.  I'm willing to be convinced this isn't  
the case, but personally I certainly wouldn't code a large amount of  
hours based on the say-so of people I'd never worked with before.   
Perhaps some sort of metric could be created for more reliable payers,  
like a rating system of integrity?

- Agreement of project goals.  Who defines the project?  Who gets what  
they want?  Based on money?  Based on some arbitration?  What and who  
defines "success"?

- Corporate structure for payments.  If there is an agent in between  
the coders and the funders, then what kind of agent is that?  For- 
profit?  Not-for-profit?  Who pays for the creation of this entity?   
It's possibly the case that Digium Inc. is not the best place for this  
funding repository, though possibly that would make life a lot easier  
from an organizational standpoint.  (not sure about taxes, though.)

- How to pay?  Obviously, the more the merrier, but credit cards, bank  
accounts, PayPal, and other payment instruments are complex and  
expensive.  Payment to consultants is another problem - taxation may  
be a problem again.

- Serious interest.  This has been a topic of conversation for the  
last 6 years that I'm aware of, and none of the concepts or problems  
I'

Re: [asterisk-users] Dealing with progress codes

[Nathan Bowyer]

On Thu, Oct 30, 2008 at 1:40 PM, Juan Rodríguez <[EMAIL PROTECTED]> wrote:

> With a script connecting to a DB server and looking for the prefix, is a
> good solution. This way you don't need to force the user to dial the the
> leading 1 (or not to do it), you just look on the DB server and if it does
> not matches a local prefix then you dial with the leading 1.


What method do you use to determine if an area code or NPA-NXX is local to
your area?  In my area, with the presence of things like EACS extended
calling or other methods, I haven't found any reliable method for this.

In case EACS is something specific to my area, its a line-based feature that
extends a number's calling scope, but seems to be different from metro
calling and not specifically defined anywhere but Telcordia documents, I
suppose.
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Re: [asterisk-users] Asterisk Legacy PBX

[Mark Best]

If I were to guess (with no config files it's really just a guess). I
would think your Dial-plan logic isn't using the right 'trunk group' for
calls.

 

 

context=from-pstn

group=0

 

context=from-legacy

group=4

 

 

[from-pstn]

exten => _.,1,Dial(Zap/g4/${EXTEN},190,r)

exten => _.,n,Hangup()

 

[from-legacy]

exten => _.,1,Dial(Zap/g0/${EXTEN},190,r)

exten => _.,n,Hangup()

 

 

Please post your configs.

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sriram
Sent: Thursday, October 30, 2008 10:23 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk Legacy PBX

 

Hi All

 

I am trying to setup :

 

PSTN E1 ---> Asterisk-->Legacy PBX--->Legacy Analog extensions.

 

I've followed steps using  :
http://www.voipinfo.org/wiki/view/Asterisk-Panasonic

 

i get the green light (sync) on both the 2nd span of digium TE420P (that
is cnnected to the legacy pbx pri card) and the pri card of the legacy
pbx. but when i try to make a call to asterisk so that it can send the
call to the legacy pbx using Dial command - it exits saying -
CHANUNAVAIL , but if i try to dial an external PSTN number the call gets
thru..

 

Any help apprecriated.

 

Thnks

Sriram

 

 
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[asterisk-users] Music On Hold (from a Sound card) Help

[Timothy Smith]

Hi,

I would like to get musiconhold from a sound card. This is because I want to
kind of be a DJ and easily change the music playing, etc. However, I
followed the instructions at
http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf but no
success. i have

[mycustom]
mode=custom
directory=/var/lib/asterisk/mohmp3
application=/usr/sbin/ast-playlinein

and =/usr/sbin/ast-playlinein contains

#!/bin/bash
/usr/bin/arecord -q -c 1 -r 8000 --buffer-size=2048 -f S16_LE -t raw

Am running asterisk as root. When i'm playing a music file (using amarok),
my music onhold is silent. Is there anything I can do?

Any help of pointers will be appreciated.

Regards,
Tim
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Re: [asterisk-users] Dealing with progress codes

[Juan Rodríguez]

1500 prefixes is not a big number. You can use a little script for it (less
than 50 lines).
With a script connecting to a DB server and looking for the prefix, is a
good solution. This way you don't need to force the user to dial the the
leading 1 (or not to do it), you just look on the DB server and if it does
not matches a local prefix then you dial with the leading 1.

I do not recommend you having all that prefixes on the Dialplan, because it
is difficult to maintain.


On Thu, Oct 30, 2008 at 10:14 AM, arkda <[EMAIL PROTECTED]> wrote:

> Thanks for the reply!
>
> Generally that's what I do, script local area codes and prefixes so that
> dialing 1 is necessary only for long distance calls. The problem here is
> that there are over 1500 area codes and prefixes (DC area) that are required
> by the carrier to not be dialed with a 1 (ie, local calls). If push comes to
> shove, I'll implement them all in the dialplan, but this just seems like a
> poor way of handling it.
>
> Running another Dial statement after a timeout with r will put the possible
> wait time close to a minute if the callee doesn't answer and voicemail picks
> up (I try to estimate as long as 30 seconds for voicemail pickup). I may
> turn this on for a temporary fix, but it's not an acceptable solution long
> term.
>
>
> On Thu, Oct 30, 2008 at 12:08 AM, Juan Rodríguez <[EMAIL PROTECTED]>wrote:
>
>> Arka:
>> I thought you would reroute the call with (or without) the leading one,
>> so, just Dial again.
>>
>> This will work and your users wont notice a BIG difference if the call is
>> answered. The problem is if the call is not answer, because if you have a
>> busy number, then your users will get something like "ring, ring...ring,
>> beep,beep...".
>>
>> For a better solution I would recommend you to get at least your local
>> prefixes and use the correct dial string with patterns. This can be achieved
>> with a script.
>>
>>
>> On Wed, Oct 29, 2008 at 6:15 PM, arkda <[EMAIL PROTECTED]> wrote:
>>
>>> I left something out on that last message, sorry.
>>>
>>> With r, not R, it will mask the message with ringing. I could then fail
>>> it over to another dial out, however from testing I've found that my users
>>> expect something to happen within 30 seconds (voicemail, pickup, etc.) The
>>> worse-case scenario would be using r a time of 60 seconds. I've been
>>> thinking of implementing this as a temp fix, but not something I want to
>>> leave in place.
>>>
>>>
>>>
>>> On Wed, Oct 29, 2008 at 5:46 PM, arkda <[EMAIL PROTECTED]> wrote:
>>>
 Thanks for the reply!

 I've played around with R to solve this (probably should have mentioned
 that), however I wasn't able to make it work. The message is still played
 (this message is from the provider). It will move to the next line in the
 dialplan, but as soon as users hear the message they hang up.

 Since the progress code comes before actual audio is played to the
 caller there has to be a way of catching this and dealing with it in the
 dialplan, but nothing I've tried so far works.


 On Wed, Oct 29, 2008 at 12:25 AM, Juan Rodríguez <[EMAIL PROTECTED]>wrote:

> Try using a R or r on the Dial command, the R option is better for you
> in my opinion.
> i.e Dial(Zap/G2/1${EXTEN},30,R) or Dial(Zap/G2/1${EXTEN}|30|R)
>
> The R option is going to generate a ring tone when the callee indicates
> ringing and is going wait for an Answer. As Progress is just for early
> media, you wont get that message.
>
> For more info on the Dial command see:
>
> http://www.voip-info.org/wiki-Asterisk+cmd+Dial
>
>
>
> On Tue, Oct 28, 2008 at 6:56 PM, arkda <[EMAIL PROTECTED]> wrote:
>
>> Some additional information.
>>
>> I played with ${DIALEDSTATUS} in place of ${HANGUPCAUSE} and got an
>> unusual result:
>>
>> [Oct 28 16:50:54] WARNING[17503]: chan_sip.c:1950 retrans_pkt: Maximum
>> retries exceeded on transmission
>> NzJlOWI0NjI5NTMwMmEwZTExYzZiZTM5YWY4MDk0MzA. for seqno 2 (Critical 
>> Response)
>>
>> This occurs about a second after the user hangs up on the error
>> message being played from the provider. I have a feeling it's trying to
>> execute the next step in the dialplan but unable since the caller hung 
>> up.
>>
>> Thoughts, criticism, insults all welcome!
>>
>>
>> On Tue, Oct 28, 2008 at 12:53 PM, arkda <[EMAIL PROTECTED]>wrote:
>>
>>> Hi,
>>>
>>> I've ran into an issue with a PRI provider in a major metropolitan
>>> area that I haven't needed to deal with before and I was hoping someone
>>> might have some insight on how to handle this within the Asterisk 
>>> dialplan.
>>>
>>> At this location users can't always tell if a number is long distance
>>> or not (there are a lot of area codes and prefixes in the vicinity).
>>> Additionally, users are required by the provide

Re: [asterisk-users] CHANUNAVAIL with a TDM800 card

[Tzafrir Cohen]

On Thu, Oct 30, 2008 at 11:03:03AM -0700, hin lee wrote:
> I got this working.  For what it's worth, here's what the issue.  
> 
> The channel wasn't getting created under FreePBX via script.  Here's what I 
> needed to do:
> 
> 1) Run genzaptelconf  to generate the zaptel configs

This generates you /etc/zaptel.conf and
/etc/asterisk/zapata-channels.conf (or /etc/asterisk/zapata-auto.conf ,
in the modified versions by some distributions).

> 
> 2) find the channel the port(s) is on.
> 
>   cat /proc/zaptel/*
>   
> 3) comment out the unused ports in /etc/zaptel.conf based on step 2 result.

Why?

> 
> 4) put in the available channel in /etc/asterisk/zapata-channels.conf
> 
> ie. channel => 1

So I gather it has generated for you zapata-auto.conf but zapata.conf
#include-s zapata-channels.conf . You're confused (or someone did some
bad integration work).

> 
> 5) comment out the unused channels in /etc/asterisk/zapata-auto.conf
> 
> 
> http://www.freepbx.org/support/documentation/administration-guide/interfacing-to-a-pstn

This thing should be removed. It is aufully confusing and completely
outdated. I think it is slightly worse than no information at all, as
the defaults of freepbx would have worked for you if you just ran
genzaptelconf .

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] CHANUNAVAIL with a TDM800 card

[hin lee]

I got this working.  For what it's worth, here's what the issue.  

The channel wasn't getting created under FreePBX via script.  Here's what I 
needed to do:

1) Run genzaptelconf  to generate the zaptel configs

2) find the channel the port(s) is on.

cat /proc/zaptel/*

3) comment out the unused ports in /etc/zaptel.conf based on step 2 result.

4) put in the available channel in /etc/asterisk/zapata-channels.conf

ie. channel => 1

5) comment out the unused channels in /etc/asterisk/zapata-auto.conf


http://www.freepbx.org/support/documentation/administration-guide/interfacing-to-a-pstn


Hope this will help the next person who may encounter this issue.


--- On Tue, 10/28/08, hin lee <[EMAIL PROTECTED]> wrote:

> From: hin lee <[EMAIL PROTECTED]>
> Subject: [asterisk-users] CHANUNAVAIL with a TDM800 card
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Date: Tuesday, October 28, 2008, 5:02 PM
> Hi,
> 
> A newbie here trying to learn Asterisk.  I've installed
> PiAF v.1.3(PBX in A Flash) and trying to set up the TDM808E
> card as a test.  For now I only have one analog line.  I
> went into the FreePBX interface and created a ZAP trunk with
> 1 as the Zap Identifier.  
> 
> When I try to call out, I get the error
> "CHANUNAVAIL" and the "All Circuits are
> busy" message.  If I try to call in, Asterisk
> doesn't pick up.  
> 
> What am I doing wrong?? I been at this for days now and I
> don't see where the issue is at!  
> 
> TIA!
> 
> --
> Output from Asterisk CLI
> --
> -- Executing [EMAIL PROTECTED]:19]
> Dial("SIP/5134-088a9c90",
> "ZAP/1/7593548|300|") in new stack
>   == Everyone is busy/congested at this time (1:0/0/1)
> -- Executing [EMAIL PROTECTED]:20]
> Goto("SIP/5134-088a9c90",
> "s-CHANUNAVAIL|1") in new stack
> -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
> -- Executing [EMAIL PROTECTED]:1]
> GotoIf("SIP/5134-088a9c90",
> "1?noreport") in new stack
> -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)
> -- Executing [EMAIL PROTECTED]:3]
> NoOp("SIP/5134-088a9c90", "TRUNK Dial failed
> due to CHANUNAVAIL (hangupcause: 0) - failing through to
> other trunks") in new stack
> -- Executing [EMAIL PROTECTED]:5]
> Macro("SIP/5134-088a9c90", "outisbusy|")
> in new stack
> -- Executing [EMAIL PROTECTED]:1]
> Playback("SIP/5134-088a9c90",
> "all-circuits-busy-now|noanswer") in new stack
> --  Playing
> 'all-circuits-busy-now' (language 'en')
>   == Spawn extension (macro-outisbusy, s, 1) exited
> non-zero on 'SIP/5134-088a9c90' in macro
> 'outisbusy'
>   == Spawn extension (macro-outisbusy, s, 1) exited
> non-zero on 'SIP/5134-088a9c90'
> 
> 
> 
> [EMAIL PROTECTED]:/etc/asterisk $ ztcfg -v
> 
> Zaptel Version: 1.4.12.1
> Echo Canceller: MG2
> Configuration
> ==
> 
> 
> Channel map:
> 
> Channel 01: FXS Kewlstart (Default) (Slaves: 01)
> Channel 02: FXS Kewlstart (Default) (Slaves: 02)
> Channel 03: FXS Kewlstart (Default) (Slaves: 03)
> Channel 04: FXS Kewlstart (Default) (Slaves: 04)
> Channel 05: FXS Kewlstart (Default) (Slaves: 05)
> Channel 06: FXS Kewlstart (Default) (Slaves: 06)
> Channel 07: FXS Kewlstart (Default) (Slaves: 07)
> Channel 08: FXS Kewlstart (Default) (Slaves: 08)
> 
> 8 channels to configure.
> 
> 
> -
> [EMAIL PROTECTED]:/etc $ cat /proc/zaptel/*
> --
> Span 1: WCTDM/0 "Wildcard TDM800P Board 1"
> (MASTER)
> IRQ misses: 1
> 
>1 WCTDM/0/0 FXSKS
>2 WCTDM/0/1 FXSKS RED
>3 WCTDM/0/2 FXSKS RED
>4 WCTDM/0/3 FXSKS RED
>5 WCTDM/0/4 FXSKS RED
>6 WCTDM/0/5 FXSKS RED
>7 WCTDM/0/6 FXSKS RED
>8 WCTDM/0/7 FXSKS RED
> 
> 
> 
> /etc/asterisk/zapata.conf
> 
> ;
> ; Zapata telephony interface
> ;
> ; Configuration file
> 
> [trunkgroups]
> 
> [channels]
> 
> language=en
> context=from-zaptel
> signalling=fxs_ks
> rxwink=300  ; Atlas seems to use long (250ms)
> winks
> ;
> ; Whether or not to do distinctive ring detection on FXO
> lines
> ;
> ;usedistinctiveringdetection=yes
> 
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=no
> echotraining=800
> rxgain=0.0
> txgain=0.0
> group=0
> callgroup=1
> pickupgroup=1
> immediate=no
> 
> ;faxdetect=both
> faxdetect=incoming
> ;faxdetect=outgoing
> ;faxdetect=no
> 
> ;Include genzaptelconf configs
> #include zapata-channels.conf
> 
> ;Include AMP configs
> #include zapata_additional.conf
> 
> 
> /etc/zaptel.conf
> 
> # Autogenerated by /usr/local/sbin/genzaptelconf 

Re: [asterisk-users] [SOLVED] SIP # DTMF

[Rodolfo Alcazar Portillo]

Am Donnerstag, den 30.10.2008, 12:17 -0400 schrieb Rodolfo Alcazar
Portillo:
> Hi. In creating a custom extension, and dialing
> SIP/222/333#444, seems the party receives only "333"

Solved, the problem was on my SPA3102, old dialplan:

(**|*x.|x.|**x.)

and now:

(**|*x.|x.|**x.|xxx#x.)

Thanks!
-- 
Rodolfo Alcazar
Responsable red y datos

Deutsche Gesellschaft für
Technische Zusammenarbeit (GTZ) GmbH

Programa de Apoyo a la Gestión Pública Descentralizada y
Lucha Contra La Pobreza - PADEP
Av. Sánchez Lima 2226
La Paz, Bolivia

Tel: +591 22417628 (121)
Fax: +591 22417628 (126)
Web: www.padep.org.bo
Email: [EMAIL PROTECTED]


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[asterisk-users] Asterisk Legacy PBX

[Sriram]

Hi All

I am trying to setup :

PSTN E1 ---> Asterisk-->Legacy PBX--->Legacy Analog extensions.

I've followed steps using  : 
http://www.voipinfo.org/wiki/view/Asterisk-Panasonic

i get the green light (sync) on both the 2nd span of digium TE420P (that is 
cnnected to the legacy pbx pri card) and the pri card of the legacy pbx. but 
when i try to make a call to asterisk so that it can send the call to the 
legacy pbx using Dial command - it exits saying - CHANUNAVAIL , but if i try to 
dial an external PSTN number the call gets thru..

Any help apprecriated.

Thnks
Sriram

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[asterisk-users] Other lists

[Adam Moffett]

Does anybody know of a mailing list devoted to SIP device or ATA 
issues?  This is a pretty high traffic list and I'd like to not clutter 
any more than I have to.  Is there a polycom list for example?

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Re: [asterisk-users] Old mantis e-mails

[Anthony Messina]

On Thursday 30 October 2008 11:57:29 am Daniel Hazelbaker wrote:
> Is it just me or has mantis been holding onto old e-mail and finally  
> sending it?

i'm getting them too.  even the original "your license agreement is accepted" 
email.

-- 
Anthony -  http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E


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Re: [asterisk-users] Old mantis e-mails

[Mark Michelson]

Daniel Hazelbaker wrote:
> I am suddenly getting a bunch of OLD (as in 3-9 months old) e-mails  
> from mantis saying things like a note has been added to an issue etc.,  
> and yet the issue has not been touched in months and the "new note" it  
> is referring to is also months old.  Consequently, I never received  
> these e-mails before either.  The e-mail itself shows that  
> "carolina.digium.com" received the message "back in the day" but the  
> next hop (my server) shows todays date.
> 
> Is it just me or has mantis been holding onto old e-mail and finally  
> sending it?
> 
> Daniel
> 

Kevin Fleming noticed earlier this morning that the SMTP daemon was not 
functioning properly at carolina.digium.com, and so there was a big backlog of 
queued messages. Part of correcting the problem is that all these queued 
messages are finally being sent out.

Mark Michelson

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Re: [asterisk-users] Asterisk settings

[Steve Howes]

On 30 Oct 2008, at 15:31, michel freiha wrote:
> <"Do my work for me">

There is a lot more to system security and performance than just  
Asterisk config. Perhaps you should do some research on this. voip- 
info wiki is good.

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[asterisk-users] Old mantis e-mails

[Daniel Hazelbaker]

I am suddenly getting a bunch of OLD (as in 3-9 months old) e-mails  
from mantis saying things like a note has been added to an issue etc.,  
and yet the issue has not been touched in months and the "new note" it  
is referring to is also months old.  Consequently, I never received  
these e-mails before either.  The e-mail itself shows that  
"carolina.digium.com" received the message "back in the day" but the  
next hop (my server) shows todays date.

Is it just me or has mantis been holding onto old e-mail and finally  
sending it?

Daniel

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Re: [asterisk-users] SIP # DTMF

[Anthony Francis]

On many phones # sends the call.

Rodolfo Alcazar Portillo wrote:
> Hi. In creating a custom extension, and dialing
>
> SIP/222/333#444, seems the party receives only "333"
>
> What should I do to send the # symbol? or better, where can I find that
> syntax? Googled a lot, nothing.
>
> Thanks!
>   

-- 
Thank you and have any kind of day you want,

Anthony Francis



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Re: [asterisk-users] SIP # DTMF

[Eric "ManxPower" Wieling]

"core show application dial"  (this is the official application doc) 
Pay special attention to the D() option.



Rodolfo Alcazar Portillo wrote:
> Hi. In creating a custom extension, and dialing
> 
> SIP/222/333#444, seems the party receives only "333"
> 
> What should I do to send the # symbol? or better, where can I find that
> syntax? Googled a lot, nothing.
> 
> Thanks!

-- 
Consulting and design services for LAN, WAN, voice and data.  Based near 
Birmingham, AL.  Now accepting clients worldwide. Contact me for Tellabs 
echo canceling systems.  Also see http://www.fnords.org/skillslist.html

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[asterisk-users] SIP # DTMF

[Rodolfo Alcazar Portillo]

Hi. In creating a custom extension, and dialing

SIP/222/333#444, seems the party receives only "333"

What should I do to send the # symbol? or better, where can I find that
syntax? Googled a lot, nothing.

Thanks!
-- 
Rodolfo Alcazar
Responsable red y datos

Deutsche Gesellschaft für
Technische Zusammenarbeit (GTZ) GmbH

Programa de Apoyo a la Gestión Pública Descentralizada y
Lucha Contra La Pobreza - PADEP
Av. Sánchez Lima 2226
La Paz, Bolivia

Tel: +591 22417628 (121)
Fax: +591 22417628 (126)
Web: www.padep.org.bo
Email: [EMAIL PROTECTED]


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Re: [asterisk-users] Sendmail for Voicemail

[Olivier]

I tried successfully esnmp ...

2008/10/29 David <[EMAIL PROTECTED]>

> [EMAIL PROTECTED] wrote:
> > When I send email from my local asterisk machine, my IP address get's
> > RBL'd.
> >
>
> I use msmtp;
> http://msmtp.sourceforge.net/
>
>
>
> Here is my /etc/msmtprc
>
> account default
> host mail.bellsouth.net
> auto_from on
> maildomain bellsouth.net
> syslog LOG_MAIL
>
> /etc/asterisk/voicemail.conf
> [default]
>
> 1000 => ,David Abbott,[EMAIL PROTECTED]
>
>
>
> --
> Powered by Gentoo GNU/LINUX
> http://www.linuxcrazy.com
>
>
>
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[asterisk-users] Asterisk SVN bug segfaulting

[Julien Claassen]

hello everyone!
   I just got the newest asterisk SVN:
trunk# svnversion
152803
   and compiled it. then I made some test-calls.
1. Calling my mailbox. It worked, but quality was not good, in comparison to 
1.6.0-beta9.
   I called via mISDn.
2. Just call myself.
   Result: Ringing and asterisk segfaulting.
3. Same for calling some gtalk-number and using app Dial mISDN/1/my_number.
   I got this a few times, even when it worked (halfway):
Huh?  Child handler, but nobody there?
   I got the following in addition to an mISDN call outside:
Didn't find BC so temporarily creating dummy BC (l3id:30001) on this port.
   and:
P[ 1] Taps should be power of 2
   I don't know where to look for those "Taps" in the last message.
   I could create a full output of this and send it to someone knowledgeable 
with the mISDN part. As I said: Can't sign-in to the bugtracker due to graphic 
capture codes. Sorry for the inconvenience of that.
   Kindest regards
  Julien


Music was my first love and it will be my last (John Miles)

 FIND MY WEB-PROJECT AT: 
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
=== AND MY PERSONAL PAGES AT: ===
http://www.juliencoder.de

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[asterisk-users] Connection two asterisk via SIP (call forward)

[Frank Becker]

Hi all,

I try to connect two asterisk-server together. There is a server
(obelix) which receives a call. This call should be transfered to
another server.


In my dialplan at obelix I have the following:
exten => 920622201,1,Dial(SIP/outbound:[EMAIL PROTECTED]:${EXTEN})
exten => 920622201,n,Hangup
exten => i,1,Congestion
exten => t,1,Congestion

If I call the number 920622201 obelix shows an output like this

obelix*CLI>
  == ISDN1#02: Incoming call '2029870' -> '920622201'
-- ISDN1#02: Updated channel name: CAPI/ISDN1/920622201-43
-- Executing Dial("CAPI/ISDN1/920622201-43",
"SIP/outbound:[EMAIL PROTECTED]:920622201") in new stack
-- Called outbound:[EMAIL PROTECTED]:920622201


It seems to be correct but at the other server there is no reaction.
Nothing happens.


At the other server I have the following in the sip.conf

[outbound]
context=outbound
type=peer
secret=secretpassword
auth=plaintext
host=192.168.100.2
nat=yes


Can anybody help me?

Best regards and many thanks in advance

Frank Becker


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Re: [asterisk-users] R key with Siemens Gigaset IP (was MWI with Siemens Gigaset S450IP)

[Olivier]

2008/10/30 Alan Lord <[EMAIL PROTECTED]>

> Olivier wrote:
> 
> > Alan,
> >
> > Did you get any success with MWI ?
> > With mine, Asterisk is getting 481 replies whenever Asterisk sends
> > NOTIFY updates.
> >
> > Cheers
>
> I don't think so no.
>
> The lamp blinks when I've missed a call but I don't think it correctly
> identifies if there are messages in the * vmailbox.


Normally, you should read the unread message count.

>
>
> To be honest it hasn't troubled me that much as when a message is left
> on the * server I get an email notification too so it's not like I miss
> much :-).

Of course ...

>
>
> But it would be good to try and fix this.
>
> I get a pair of error messages like this frequently (about 1/minute I
> guess) from the * server:
>
> [Oct 30 08:42:05] WARNING[14051]: chan_sip.c:12543 handle_response:
> Remote host can't match request NOTIFY to call
> '[EMAIL PROTECTED]'. Giving up.
> [Oct 30 08:42:15] WARNING[14051]: chan_sip.c:12543 handle_response:
> Remote host can't match request NOTIFY to call
> '[EMAIL PROTECTED]'. Giving up.
>
>
> which I believe might have something to do with it?


I think I've got the same messages.
In another thread (see
http://lists.digium.com/pipermail/asterisk-users/2008-October/219511.html),
I observed that :
- first message notification (just after base station boots and registers)
works ok
- further notifications fails (481 replies)

The only difference I can see between both is that the successful one has a
Subscription-State: active field.

I didn't dig enough to know if refusing NOTIFY without Subscription-State is
legitimate nor if "forging" a NOTIFY with this Subscription-State would make
Gigaset accept new notifications.



>
>
> 10.0.0.2 is the Asterisk server.
>
> My sip.conf is like so:
>
> > [general]
> > srvlookup=yes
> > disallow=all
> > allow=alaw
> > allow=g722
> > allow=gsm
> > dtmfmode=auto
> > subscribemwi=yes
> >
> > [101]
> > type=friend
> > callerid=Alan Lord <101>
> > secret=bigsecret
> > qualify=yes ; Qualify peer is no more than 2000 ms away
> > nat=no ; This phone is not natted
> > host=dynamic ; This device registers with us
> > canreinvite=no ; Asterisk by default tries to redirect
> > context=alanl ; the internal context controls what we can do
> > mailbox=101 ; Voicemail Boxes
>
> The S685IP supports G722 which we have been testing between our offices
> (sounds great!).
>
> If anyone has any ideas that would be cool.
>
> Cheers
>
> Alan
>
>
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[asterisk-users] Asterisk settings

[michel freiha]

Dear All,
I have the below settings on my asterisk server and I need to know if there
is a any problem in a setting regarding performance or security..Please
check and let me know:


Global Settings:

  SIP Port:   5060
  Bindaddress:IP_ADDRESS
  Videosupport:   No
  AutoCreatePeer: No
  Allow unknown access:   Yes
  Allow subscriptions:Yes
  Allow overlap dialing:  Yes
  Promsic. redir: No
  SIP domain support: Yes
  Call to non-local dom.: Yes
  URI user is phone no:   No
  Our auth realm  asterisk
  Realm. auth:No
  Always auth rejects:No
  Call limit peers only:  No
  Direct RTP setup:   No
  User Agent: Asterisk PBX
  MWI checking interval:  10 secs
  Reg. context:   (not set)
  Caller ID:  asterisk
  From: Domain:
  Record SIP history: Off
  Call Events:Off
  IP ToS SIP: none
  IP ToS RTP audio:   none
  IP ToS RTP video:   none
  T38 fax pt UDPTL:   No
  RFC2833 Compensation:   No
  SIP realtime:   Enabled

Global Signalling Settings:
---
  Codecs: 0x10e (gsm|ulaw|alaw|g729)
  Codec Order:g729:20,ulaw:20,alaw:20,gsm:20
  T1 minimum: 100
  Relax DTMF: No
  Compact SIP headers:No
  RTP Keepalive:  0 (Disabled)
  RTP Timeout:0 (Disabled)
  RTP Hold Timeout:   0 (Disabled)
  MWI NOTIFY mime type:   application/simple-message-summary
  DNS SRV lookup: Yes
  Pedantic SIP support:   No
  Reg. min duration   60 secs
  Reg. max duration:  3600 secs
  Reg. default duration:  120 secs
  Outbound reg. timeout:  20 secs
  Outbound reg. attempts: 0
  Notify ringing state:   Yes
  Notify hold state:  No
  SIP Transfer mode:  open
  Max Call Bitrate:   384 kbps
  Auto-Framing:   No
localhost*CLI>
Default Settings:
-
  Context:default
  Nat:RFC3581
  DTMF:   rfc2833
  Qualify:0
  Use ClientCode: No
  Progress inband:Never
  Language:   (Defaults to English)
  MOH Interpret:  default
  MOH Suggest:
  Voice Mail Extension:   asterisk

Realtime SIP Settings:
--
  Realtime Peers: Yes
  Realtime Users: Yes
  Cache Friends:  No
  Update: Yes
  Ignore Reg. Expire: No
  Save sys. name: No
  Auto Clear: 120

Thanks a lot
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Re: [asterisk-users] up to 3000 lines capacity asterisk Deployment

[Andrew Latham]

Cost per analog port can be quite low if done rightl.  I would look at
using Xorcom's 32 port asterisk appliances with additional Astribanks
plugged in and a "switch" or "dundi" setup.



On Thu, Oct 30, 2008 at 5:02 AM, Dumpolid Exeplish <[EMAIL PROTECTED]> wrote:
> Hello All,
>
> I have a request from a prospectieve client to deploy a PBX capacity
> that can do up to 3000+ lines within a geographic region similar to a
> campus. The client wants analog lines for extensions and maybe VoIP
> for some backhaul traffic while the other traffic would be carrid via
> E1 channels. The client has other proposals to buy a mid, range telco
> switch from alcatel or simens but i am trying to convince him
> otherwise. Though i have deployed Asterisk PBX in mid range offices
> (like up to 60 lines), i have never deployed anything of this
> magnitude. Can anyone help me by giving me information on how this
> sort of deployment would work, keeping in mind that the extensions
> would all be analog and not VOIP. I have seen some documentation on
> SpiderMUX (TDMoE) and some other channelized MUXes using USB
> interfaces but then, how would i even start to build a PBX of this
> magnitude in the first instance? Any suggestions would be welcomed
>
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-- 
Andrew "lathama" Latham

TuxTone Inc.
http://TuxTone.com
[EMAIL PROTECTED]

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Re: [asterisk-users] network design philosophy and practice

[Andrew Latham]

Alex

I see a fair bit of separate physical networks because of different
management of phones vs IT.  In the old businesses Facilities handles
the communications and IT is playing catchup all the time

So in these businesses where the IT side is swapping switches on a
weekly basis it is safer to have a separate physical network.


Andrew


On Wed, Oct 29, 2008 at 11:30 AM, Alex Balashov
<[EMAIL PROTECTED]> wrote:
> I'm pretty sure they meant two logical networks.  At least, I hope they did.
>
> David Gibbons wrote:
>
>> Two separate networks? Did I miss something? I feel like I'm taking crazy 
>> pills! Two separate physical networks means twice the hassle, twice the 
>> maintenance, twice the cost, twice the headache. Not to mention the fact 
>> that the whole idea of VOIP is to simplify IT and focus on converging data 
>> and voice networks.
>>
>> This is what VLANs and QOS do best. I dare say it's what they were designed 
>> foe. I can't think of any reason that I would ever recommend two ports per 
>> desk to support telephony -- ever. It's ludicrous to think that two ports 
>> will be better than one if we're setting up our VLANs and QOS properly. A 
>> phone takes very, very little bandwidth away from the desktop and a decent 
>> one will support tagging its frames for the alternate voice VLAN.
>>
>> --snip--
>> In almost all cases it is much better to have two seperate networks.
>> This may be impractical in some smaller installs, but in any office
>> setting we always do this. The only reason I can think of not to is to
>> eliminate the cost of the second cable.
>> --snip--
>>
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>
> --
> Alex Balashov
> Evariste Systems
> Web: http://www.evaristesys.com/
> Tel: (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (706) 338-8599
>
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-- 
Andrew "lathama" Latham

TuxTone Inc.
http://TuxTone.com
[EMAIL PROTECTED]

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Re: [asterisk-users] Sangoma Question

[BJ Weschke]

Jeremy Mann wrote:
>
> Any advise on troubleshooting this:
>
> Oct 29 02:25:58 nurscarepbx kernel: wanpipe2: OOF alarm is OFF
>
> Oct 29 02:25:58 nurscarepbx kernel: wanpipe2: RED alarm is OFF
>
> Oct 29 02:26:05 nurscarepbx kernel: wanpipe2: RAI alarm is OFF
>
> Oct 29 02:26:05 nurscarepbx kernel: wanpipe2: RED alarm is OFF
>
> It happens nightly, and I have to reset asterisk to “clear” it. 
> Zap/Dahdi channels wont’ work until I do.
>
 The message is that you're losing frame/timing on your circuit. So there's two 
issues really. The first is that chan_zap/dahdi using Sangoma wanpipe drivers 
isn't recovering from a red alarm. The second is trying to find out why your 
provider is dropping frame on you every night. 

 You'll want to consult Sangoma support for the first issue, and call your 
carrier on the second one.

--
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http://www.btwtech.com/




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Re: [asterisk-users] Asterisk and rawplayer

[BJ Weschke]

Ade Vickers wrote:
>> -Original Message-
>>
>> Hi Folks, 
>>
>> I'm using the "rawplayer" program to provide my 
>> music-on-hold, and it works very well, with one small 
>> drawback: every time I reset Asterisk, for any reason, the 
>> MoH resets to the beginning of the list. Since MoH isn't used 
>> that often, it basically means the same track is played over 
>> & over again...
>>
>> What I'd like to do is have rawplayer continuously playing 
>> away in the background, even if it's playing to itself only, 
>> so there's an excellent chance that any caller who will be 
>> given the pleasure of my MoH choices, will get a different 
>> tune to the one s/he heard last time...
>>
>>
>> Any ideas?
>>
>>
>> Asterisk is v1.4.18.1, running on Ubuntu 2.6.20-15.27-server.
>>
>>
>> 
>
> I'm still stuck with this, and would appreciate any thoughts...
>
> Thanks in advance!
> Ade.
>
>
>
>   
 This would probably involve some kind of IPC named pipe or other inter process 
method of getting the data from pt A to pt B to work.  While technically 
possible, it's not a trivial amount of work to get it going in the codebase. 
You might be better off with something like streaming MP3 over http or 
something else like that if you're looking for something with no code 
modifications. 

 Are you really resetting Asterisk that much that this becomes a problem? If 
so, why?
 

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




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Re: [asterisk-users] Dealing with progress codes

[arkda]

Thanks for the reply!

Generally that's what I do, script local area codes and prefixes so that
dialing 1 is necessary only for long distance calls. The problem here is
that there are over 1500 area codes and prefixes (DC area) that are required
by the carrier to not be dialed with a 1 (ie, local calls). If push comes to
shove, I'll implement them all in the dialplan, but this just seems like a
poor way of handling it.

Running another Dial statement after a timeout with r will put the possible
wait time close to a minute if the callee doesn't answer and voicemail picks
up (I try to estimate as long as 30 seconds for voicemail pickup). I may
turn this on for a temporary fix, but it's not an acceptable solution long
term.

On Thu, Oct 30, 2008 at 12:08 AM, Juan Rodríguez <[EMAIL PROTECTED]> wrote:

> Arka:
> I thought you would reroute the call with (or without) the leading one, so,
> just Dial again.
>
> This will work and your users wont notice a BIG difference if the call is
> answered. The problem is if the call is not answer, because if you have a
> busy number, then your users will get something like "ring, ring...ring,
> beep,beep...".
>
> For a better solution I would recommend you to get at least your local
> prefixes and use the correct dial string with patterns. This can be achieved
> with a script.
>
>
> On Wed, Oct 29, 2008 at 6:15 PM, arkda <[EMAIL PROTECTED]> wrote:
>
>> I left something out on that last message, sorry.
>>
>> With r, not R, it will mask the message with ringing. I could then fail it
>> over to another dial out, however from testing I've found that my users
>> expect something to happen within 30 seconds (voicemail, pickup, etc.) The
>> worse-case scenario would be using r a time of 60 seconds. I've been
>> thinking of implementing this as a temp fix, but not something I want to
>> leave in place.
>>
>>
>>
>> On Wed, Oct 29, 2008 at 5:46 PM, arkda <[EMAIL PROTECTED]> wrote:
>>
>>> Thanks for the reply!
>>>
>>> I've played around with R to solve this (probably should have mentioned
>>> that), however I wasn't able to make it work. The message is still played
>>> (this message is from the provider). It will move to the next line in the
>>> dialplan, but as soon as users hear the message they hang up.
>>>
>>> Since the progress code comes before actual audio is played to the caller
>>> there has to be a way of catching this and dealing with it in the dialplan,
>>> but nothing I've tried so far works.
>>>
>>>
>>> On Wed, Oct 29, 2008 at 12:25 AM, Juan Rodríguez <[EMAIL PROTECTED]>wrote:
>>>
 Try using a R or r on the Dial command, the R option is better for you
 in my opinion.
 i.e Dial(Zap/G2/1${EXTEN},30,R) or Dial(Zap/G2/1${EXTEN}|30|R)

 The R option is going to generate a ring tone when the callee indicates
 ringing and is going wait for an Answer. As Progress is just for early
 media, you wont get that message.

 For more info on the Dial command see:

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



 On Tue, Oct 28, 2008 at 6:56 PM, arkda <[EMAIL PROTECTED]> wrote:

> Some additional information.
>
> I played with ${DIALEDSTATUS} in place of ${HANGUPCAUSE} and got an
> unusual result:
>
> [Oct 28 16:50:54] WARNING[17503]: chan_sip.c:1950 retrans_pkt: Maximum
> retries exceeded on transmission
> NzJlOWI0NjI5NTMwMmEwZTExYzZiZTM5YWY4MDk0MzA. for seqno 2 (Critical 
> Response)
>
> This occurs about a second after the user hangs up on the error message
> being played from the provider. I have a feeling it's trying to execute 
> the
> next step in the dialplan but unable since the caller hung up.
>
> Thoughts, criticism, insults all welcome!
>
>
> On Tue, Oct 28, 2008 at 12:53 PM, arkda <[EMAIL PROTECTED]> wrote:
>
>> Hi,
>>
>> I've ran into an issue with a PRI provider in a major metropolitan
>> area that I haven't needed to deal with before and I was hoping someone
>> might have some insight on how to handle this within the Asterisk 
>> dialplan.
>>
>> At this location users can't always tell if a number is long distance
>> or not (there are a lot of area codes and prefixes in the vicinity).
>> Additionally, users are required by the provider to dial the full 10 
>> digit
>> number even if a call is local since a local call could be for a few
>> different area codes and prefixes. The problem is the provider requires 
>> a 1
>> in front of the number for long distance calls, but errors out if the 
>> call
>> has a 1 in front and the call is local.
>>
>> As a result, users are complaining that they are constantly having to
>> redial with or without the 1. I've tracked down this behavior when a call
>> fails:
>>
>> -- Executing [EMAIL PROTECTED]:1] Set("SIP/user9-b696fb58",
>> "GROUP(default)=dialpool") in new stack
>> -- Executin

Re: [asterisk-users] XML Cisco config file

[César García]

OK done:

http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP#Cisco7911Gfirmware841SR1Sconfigurationex

2008/10/29 OCG Technical Support <[EMAIL PROTECTED]>

>  Post it on the wiki!  I'm sure I'll need it someday
>
>
>
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *César García
> *Sent:* October 29, 2008 6:54 PM
> *To:* Asterisk Users List
> *Subject:* Re: [asterisk-users] XML Cisco config file
>
>
>
> Well guys I got it, I started up again making the xml file according to
> this:
>
> http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP#Downgradingthefirmware
>
> And... voila ! 7911G working with Asterisk and firmware 8.4.0!!! if anybody
> need the xml, let me know :)
>
> 2008/10/28 Lincoln King-Cliby <[EMAIL PROTECTED]>
>
> I'm not sure if it's the only issue but you're going to have issues with
>
>
>
> **Etiqueta_del_telefono**
>
>
>
> The text within the  tag is a maximum of 11 or 12 characters (I
> can't remember off the top of my head), if it's longer than that--I count 21
> characters in the example, the phone will reject the entire configuration
> file more or less silently (it is logged in the phone's debug log at 
> http:// ip address>/ but there's no display on the phone itself).
>
>
>
> That sounds like at least part of what's happening in your case.
>
>
>
> --
>
> Lincoln King-Cliby, CTS
>
> Applications Engineer
>
> ControlWorks Consulting, LLC
>
> http://www.controlworks.com
>
> Crestron Authorized Independent Programmer
>   --
>
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *César García
> *Sent:* Tuesday, October 28, 2008 6:46 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] XML Cisco config file
>
>
>
> Hello guys, anybody here that can help me checking out this xml file, cause
> I am traying to configure some cisco 7911G phones to asterisk and I can't
> get it done
>
> thanks
>
> a paste of the file is here:
>
> http://pastebin.ca/1239083
>
> --
> http://celord.blogspot.com/
>
>
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>
>
>
> --
> http://celord.blogspot.com/
>
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-- 
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Re: [asterisk-users] Linux Kernel >=2.6.25 Realtime issues

[Sven Geggus]

Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
>> I'm not using a Debian Kernel, thus this looks like a problem of a
>> particular Kernel Option.
> 
> Do you use debs?

vanilla kernel (currently 2.6.27.4) and debian kernel-packages to build. I
have now Idea, if my machine will work with the official lenny kernel, but I
can give it a try.

> source packages?

vanilla kernel and zaptel package build with make-kpkg:
linux-image-2.6.27.4_XXX_amd64.deb
zaptel-modules-2.6.27.4_1.4.11~dfsg-2+XXX_amd64.deb

> Any chance you could test this with the official kernel?

> What kernel config do you use?

http://home.geggus.net/pub/config-2.6.27.4

Sven

-- 
"linux is evolution, not intelligent design"
(Linus Torvalds)

/me is [EMAIL PROTECTED], http://sven.gegg.us/ on the Web


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[asterisk-users] Sangoma Question

[Jeremy Mann]

Any advise on troubleshooting this:

Oct 29 02:25:58 nurscarepbx kernel: wanpipe2: OOF alarm is OFF
Oct 29 02:25:58 nurscarepbx kernel: wanpipe2: RED alarm is OFF
Oct 29 02:26:05 nurscarepbx kernel: wanpipe2: RAI alarm is OFF
Oct 29 02:26:05 nurscarepbx kernel: wanpipe2: RED alarm is OFF

It happens nightly, and I have to reset asterisk to "clear" it.  Zap/Dahdi 
channels wont' work until I do.

Jeremy Mann
Director of IT
Texas Health Management Group
Direct Line: 817-310-4956
Main Line: 817-310-4999
Fax: 817-310-4990
Email: [EMAIL PROTECTED]



This e-mail, facsimile, or letter and any files or attachments transmitted with 
it contains information that is confidential and privileged. This information 
is intended only for the use of the individual(s) and entity(ies) to whom it is 
addressed. If you are the intended recipient, further disclosures are 
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any disclosure, copying, printing, or use of this information is strictly 
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Re: [asterisk-users] Current Open Source Billing Package

[Outback Dingo]

also check jBilling, its a telco oriented billing system, quite nice, there
is also freeside... it might work out for you

On Thu, Oct 30, 2008 at 3:49 PM, Stephen Wingfield <[EMAIL PROTECTED]> wrote:

>
> - Original Message -
> From: "Jerry Jones" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Wednesday, October 29, 2008 6:55 PM
> Subject: [asterisk-users] Current Open Source Billing Package
>
>
> > After spending a couple hours scanning for an open source (non-
> > commercial) billing package yesterday I am underwhelmed. Almost all of
> > the packages listed on the WIKI appear to be defunct, for several
> > years now. I will be happy to get a login and edit them out if that is
> > the proper method to do so.
> >
> > My requirements are very minimal and at this point unless I have
> > missed something will just write my own.
> >
> > I do not do calling cards. I have no near term need for the package to
> > actually talk with asterisk at all, other than to import the CDR
> > either via files or as a login to MySQL.
> >
> > I do have monthly recurring charges which need to be included monthly.
> >
> > I do occasionally have need to one off (manual) billing charges.
> >
> > Rating for calls would be nice but not mandatory ( we have very
> > minimal International).
> >
> > Ability to export to an accounting package a plus.
> >
> > Ability to generate hard copy Invoices and/or email them to the cust.
> >
> > Ability to generate a list of current Invoices.
> >
> > Runs on Linux.
> >
> > All in all not a very complex set of requirements, but the few
> > packages that seem to be currently offered generally do not fit the
> > bill. Yes there are many commercial packages, but unless they are very
> > minimal in cost I have no interest in them.
> >
> > So my question is, have a missed a golden nugget out there?
> >
> >
> > tia
> > Jerry
>
> Jerry
>
> Our telcoware software can provide you all you wish and more.
> This is a commercially supported package.
> Please contact me offline for all details.
>
> steve 'at} bicomsystems d*t c*m
> www.bicomsystems.com
>
>
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Re: [asterisk-users] Linux Kernel >=2.6.25 Realtime issues

[Tzafrir Cohen]

On Thu, Oct 30, 2008 at 11:36:26AM +, Sven Geggus wrote:
> Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> 
> > Not here (Debian Lenny, kernel 2.6.26-1-amd64, official asterisk
> > packages 1:1.4.21.2~dfsg-2 ).
> > 
> > Can you please test those? (and report a major bug if this is
> > reproducable with them as well)
> 
> I'm not using a Debian Kernel, thus this looks like a problem of a
> particular Kernel Option.

Do you use debs? source packages?

Any chance you could test this with the official kernel? What kernel
config do you use?

> 
> Are you using ztdummy?

Right now yes. I also used an hardware Zaptel device a while ago.

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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[asterisk-users] Trouble with SIP/NAT

[Noro Hasina]

hello !
If I want to use SIP instead of IAX protocol, how can I resolve the NAT
problem with SIP?
Thank you for answering!
Sincerly
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Re: [asterisk-users] SIP REGISTER

[Alex Balashov]

By default, the interval at which the qualify pings are sent is, indeed 
quite low.

There is no consequence to disabling it except for the obvious 
implication that Asterisk then has no way way of knowing if the peer is 
dead without first trying to reach it, every time and with every request.

But there are disadvantages to using 'qualify' for that purpose, too; 
sometimes there is arbitrary latency in the network that can cause peers 
to become marked 'Unavailable' rather whimsically.

The answer is basically: do whatever you want.  No "best practices" here.

Personally, I'd recommend a qualify setting like 2000.


michel freiha wrote:

> Dear Alex,
> 
> The problem is that the asterisk server is sending these packets 
> continuously with no stop and with a negligible duration between packets 
> for the same extension...My Asterisk server read the extensions from the 
> database and not from extensions.conf...There is a field in the sip 
> buddies table with name qualify and with type char...WHat do you suggest 
> me to do? Put the value or qualify to no or ichange the type to int and 
> put a numeric value inside?
> 
> If I put the value to no what this the disadvantages?
> 
> Regards
> 
> On Thu, Oct 30, 2008 at 1:30 PM, Alex Balashov 
> <[EMAIL PROTECTED] > wrote:
> 
> These are requests where one endpoint "pings" the other to check if it
> is still alive.
> 
> What is the problem?
> 
> michel freiha wrote:
> 
>  > Hi all,
>  > I'm facing an issue with my asterisk server when an extension (X-Lite
>  > softphone) tries to register on it...A huge amount of packets is
>  > exchanged between endpoint and asterisk server while the X-Lite is
>  > online...Even when I sign out from X-Lite, the asterisk server
> continues
>  > sending packets to my machine...Can Someone help me in that?
> Please find
>  > the SIP packets between asterisk and X-Lite on
> http://pastebin.com/d85f913e
>  > Regards
>  >
>  >
>  >
> 
>  >
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>  >
>  > asterisk-users mailing list
>  > To UNSUBSCRIBE or update options visit:
>  >http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> --
> Alex Balashov
> Evariste Systems
> Web: http://www.evaristesys.com/
> Tel: (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (706) 338-8599
> 
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Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Linux Kernel >=2.6.25 Realtime issues

[Sven Geggus]

Tzafrir Cohen <[EMAIL PROTECTED]> wrote:

> Not here (Debian Lenny, kernel 2.6.26-1-amd64, official asterisk
> packages 1:1.4.21.2~dfsg-2 ).
> 
> Can you please test those? (and report a major bug if this is
> reproducable with them as well)

I'm not using a Debian Kernel, thus this looks like a problem of a
particular Kernel Option.

Are you using ztdummy?

Sven

-- 
"linux is evolution, not intelligent design"
(Linus Torvalds)

/me is [EMAIL PROTECTED], http://sven.gegg.us/ on the Web


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Re: [asterisk-users] SIP REGISTER

[michel freiha]

Dear Alex,

The problem is that the asterisk server is sending these packets
continuously with no stop and with a negligible duration between packets for
the same extension...My Asterisk server read the extensions from the
database and not from extensions.conf...There is a field in the sip buddies
table with name qualify and with type char...WHat do you suggest me to do?
Put the value or qualify to no or ichange the type to int and put a numeric
value inside?

If I put the value to no what this the disadvantages?

Regards

On Thu, Oct 30, 2008 at 1:30 PM, Alex Balashov <[EMAIL PROTECTED]>wrote:

> These are requests where one endpoint "pings" the other to check if it
> is still alive.
>
> What is the problem?
>
> michel freiha wrote:
>
> > Hi all,
> > I'm facing an issue with my asterisk server when an extension (X-Lite
> > softphone) tries to register on it...A huge amount of packets is
> > exchanged between endpoint and asterisk server while the X-Lite is
> > online...Even when I sign out from X-Lite, the asterisk server continues
> > sending packets to my machine...Can Someone help me in that? Please find
> > the SIP packets between asterisk and X-Lite on
> http://pastebin.com/d85f913e
> > Regards
> >
> >
> > 
> >
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>
> --
> Alex Balashov
> Evariste Systems
> Web: http://www.evaristesys.com/
> Tel: (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (706) 338-8599
>
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Re: [asterisk-users] SIP REGISTER

[Alex Balashov]

These are requests where one endpoint "pings" the other to check if it 
is still alive.

What is the problem?

michel freiha wrote:

> Hi all,
> I'm facing an issue with my asterisk server when an extension (X-Lite 
> softphone) tries to register on it...A huge amount of packets is 
> exchanged between endpoint and asterisk server while the X-Lite is 
> online...Even when I sign out from X-Lite, the asterisk server continues 
> sending packets to my machine...Can Someone help me in that? Please find 
> the SIP packets between asterisk and X-Lite on http://pastebin.com/d85f913e
> Regards
> 
> 
> 
> 
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-- 
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Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] Linux Kernel >=2.6.25 Realtime issues

[Tzafrir Cohen]

On Thu, Oct 30, 2008 at 10:55:42AM +, Sven Geggus wrote:
> Hello,
> 
> looks like Asterisk (at least the Version 1.4.21.2 from debian lenny, which
> I use) has a serious Problem with pseudo-realtime mode (-p switch) in
> conjunction with recent Kernels.
> 
> Whenn using -p Asterisk simply hangs forever on startup.

Not here (Debian Lenny, kernel 2.6.26-1-amd64, official asterisk
packages 1:1.4.21.2~dfsg-2 ).

Can you please test those? (and report a major bug if this is
reproducable with them as well)

> 
> This is almost certainly related to the CFS SCHED_FIFO changes:
> http://lwn.net/Articles/296419/
> 
> Do you know how a workaround or patch how to run asterisk in pseudo-realtime
> mode on this Kernels?
> 
> Everything works fine without using -p!
> 
> Sven
> 
> -- 
> "linux is evolution, not intelligent design"
> (Linus Torvalds)
> 
> /me is [EMAIL PROTECTED], http://sven.gegg.us/ on the Web
> 
> 
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-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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[asterisk-users] Linux Kernel >=2.6.25 Realtime issues

[Sven Geggus]

Hello,

looks like Asterisk (at least the Version 1.4.21.2 from debian lenny, which
I use) has a serious Problem with pseudo-realtime mode (-p switch) in
conjunction with recent Kernels.

Whenn using -p Asterisk simply hangs forever on startup.

This is almost certainly related to the CFS SCHED_FIFO changes:
http://lwn.net/Articles/296419/

Do you know how a workaround or patch how to run asterisk in pseudo-realtime
mode on this Kernels?

Everything works fine without using -p!

Sven

-- 
"linux is evolution, not intelligent design"
(Linus Torvalds)

/me is [EMAIL PROTECTED], http://sven.gegg.us/ on the Web


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Re: [asterisk-users] Asterisk and rawplayer

[Ade Vickers]

> -Original Message-
> 
> Hi Folks, 
> 
> I'm using the "rawplayer" program to provide my 
> music-on-hold, and it works very well, with one small 
> drawback: every time I reset Asterisk, for any reason, the 
> MoH resets to the beginning of the list. Since MoH isn't used 
> that often, it basically means the same track is played over 
> & over again...
> 
> What I'd like to do is have rawplayer continuously playing 
> away in the background, even if it's playing to itself only, 
> so there's an excellent chance that any caller who will be 
> given the pleasure of my MoH choices, will get a different 
> tune to the one s/he heard last time...
> 
> 
> Any ideas?
> 
> 
> Asterisk is v1.4.18.1, running on Ubuntu 2.6.20-15.27-server.
> 
> 

I'm still stuck with this, and would appreciate any thoughts...

Thanks in advance!
Ade.



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[asterisk-users] mp3player and shoutcast

[ACL]

My experience told me that mp3player (command) was rather unreliable with 
shoutcast. I heard nothing. If I use madplay in musiconhold.conf, everything is 
ok. But, using mp3player in extensions.conf is better than using MoH.

Pls kindly advise if we could use madplay in extensions.conf. Could we put it 
into system command?

Best Rgds,



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Re: [asterisk-users] Is anyone using * for 2 way video conferencing?

[Gordon Henderson]

On Wed, 29 Oct 2008, Jeff LaCoursiere wrote:


I've been "playing" with video phones over the past month or 2.

You've got 3 choices: Bottom-end is Xlite, etc. soft-phones.

Desktop videophones - currently Grandtream GXV3000 and ATL4000's.

Top of the range Polycom video conferencing units.

Starting with the top-of the range ones - these "just work"
Don't even need an Asterisk box. Expensive though - I did one
help setup a pair of these, one in the UK, the other
west-coast US. Both with 42" plasma screens. Very nice,
worked very well. Very expensive.


Define expensive and what models?


The softphones are free, the Grandstreams are in the £150 range, the 
ATL400's a bit more - £225ish..


I don't recall the exact model of Polycoms, but the nearest in physical 
appearance I can find now is £2600. (VSX 6000) I've a funny feeling they 
paid over £3000 each for them when they were setup which was some 3 years 
ago.


I do recall the plasma screens being just as expensive at the time.

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[asterisk-users] SIP REGISTER

[michel freiha]

Hi all,
I'm facing an issue with my asterisk server when an extension (X-Lite
softphone) tries to register on it...A huge amount of packets is exchanged
between endpoint and asterisk server while the X-Lite is online...Even when
I sign out from X-Lite, the asterisk server continues sending packets to my
machine...Can Someone help me in that? Please find the SIP packets between
asterisk and X-Lite on http://pastebin.com/d85f913e
Regards
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[asterisk-users] up to 3000 lines capacity asterisk Deployment

[Dumpolid Exeplish]

Hello All,

I have a request from a prospectieve client to deploy a PBX capacity
that can do up to 3000+ lines within a geographic region similar to a
campus. The client wants analog lines for extensions and maybe VoIP
for some backhaul traffic while the other traffic would be carrid via
E1 channels. The client has other proposals to buy a mid, range telco
switch from alcatel or simens but i am trying to convince him
otherwise. Though i have deployed Asterisk PBX in mid range offices
(like up to 60 lines), i have never deployed anything of this
magnitude. Can anyone help me by giving me information on how this
sort of deployment would work, keeping in mind that the extensions
would all be analog and not VOIP. I have seen some documentation on
SpiderMUX (TDMoE) and some other channelized MUXes using USB
interfaces but then, how would i even start to build a PBX of this
magnitude in the first instance? Any suggestions would be welcomed

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Re: [asterisk-users] Asterisk 1.6 pbx_lua not creating any contexts

[Charles Duffy]

To follow up --

pbx_lua from trunk works as advertised when backported to 1.6.

pbx_lua from asterisk 1.6 seems hopelessly broken, and I've given up
on trying to persuade it to work.

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Re: [asterisk-users] Current Open Source Billing Package

[Stephen Wingfield]

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

Sent: Wednesday, October 29, 2008 6:55 PM
Subject: [asterisk-users] Current Open Source Billing Package


> After spending a couple hours scanning for an open source (non-
> commercial) billing package yesterday I am underwhelmed. Almost all of
> the packages listed on the WIKI appear to be defunct, for several
> years now. I will be happy to get a login and edit them out if that is
> the proper method to do so.
>
> My requirements are very minimal and at this point unless I have
> missed something will just write my own.
>
> I do not do calling cards. I have no near term need for the package to
> actually talk with asterisk at all, other than to import the CDR
> either via files or as a login to MySQL.
>
> I do have monthly recurring charges which need to be included monthly.
>
> I do occasionally have need to one off (manual) billing charges.
>
> Rating for calls would be nice but not mandatory ( we have very
> minimal International).
>
> Ability to export to an accounting package a plus.
>
> Ability to generate hard copy Invoices and/or email them to the cust.
>
> Ability to generate a list of current Invoices.
>
> Runs on Linux.
>
> All in all not a very complex set of requirements, but the few
> packages that seem to be currently offered generally do not fit the
> bill. Yes there are many commercial packages, but unless they are very
> minimal in cost I have no interest in them.
>
> So my question is, have a missed a golden nugget out there?
>
>
> tia
> Jerry

Jerry

Our telcoware software can provide you all you wish and more.
This is a commercially supported package.
Please contact me offline for all details.

steve 'at} bicomsystems d*t c*m
www.bicomsystems.com 


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Re: [asterisk-users] R key with Siemens Gigaset IP (was MWI with Siemens Gigaset S450IP)

[Alan Lord]

Olivier wrote:

> Alan,
> 
> Did you get any success with MWI ?
> With mine, Asterisk is getting 481 replies whenever Asterisk sends 
> NOTIFY updates.
> 
> Cheers

I don't think so no.

The lamp blinks when I've missed a call but I don't think it correctly 
identifies if there are messages in the * vmailbox.

To be honest it hasn't troubled me that much as when a message is left 
on the * server I get an email notification too so it's not like I miss 
much :-).

But it would be good to try and fix this.

I get a pair of error messages like this frequently (about 1/minute I 
guess) from the * server:

[Oct 30 08:42:05] WARNING[14051]: chan_sip.c:12543 handle_response: 
Remote host can't match request NOTIFY to call 
'[EMAIL PROTECTED]'. Giving up.
[Oct 30 08:42:15] WARNING[14051]: chan_sip.c:12543 handle_response: 
Remote host can't match request NOTIFY to call 
'[EMAIL PROTECTED]'. Giving up.


which I believe might have something to do with it?

10.0.0.2 is the Asterisk server.

My sip.conf is like so:

> [general]
> srvlookup=yes
> disallow=all
> allow=alaw
> allow=g722
> allow=gsm
> dtmfmode=auto
> subscribemwi=yes
> 
> [101]
> type=friend
> callerid=Alan Lord <101>
> secret=bigsecret
> qualify=yes ; Qualify peer is no more than 2000 ms away
> nat=no ; This phone is not natted
> host=dynamic ; This device registers with us
> canreinvite=no ; Asterisk by default tries to redirect
> context=alanl ; the internal context controls what we can do
> mailbox=101 ; Voicemail Boxes

The S685IP supports G722 which we have been testing between our offices 
(sounds great!).

If anyone has any ideas that would be cool.

Cheers

Alan


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Re: [asterisk-users] [Kamailio-Users] [OpenSIPS-Users] Dimensioning a telephony system based on openser!

[LetMeKnow]

Hello,
I can't say the " Works only if some one is offering unmetered only service
or just doing it for fun. If it metered service like calling cards,
termination or metered DID etc, then this can be* really bad*."
If the Service providers use the Sip-B2BUA inside the Sip-Proxy servers.
then, it sounds Good.

Thanks & Regards
Ravi Prakash Sunkara
VoIP Architect & JAVA-SIP Developer
+91-882776


2008/10/30 Jai Rangi <[EMAIL PROTECTED]>

> "SIP-only accounting is "good enough" most of the time."
> Does not work in production environment. Specially when you are charging
> per second or per minute.
> Works only if some one is offering unmetered only service or just doing it
> for fun. If it metered service like calling cards, termination or metered
> DID etc, then this can be really bad.
> My 2 cents.
>
> -Jai
> "Buy unmetered SIP DID
> www.didforsale.com"
>
>
> On Wed, Oct 29, 2008 at 3:56 PM, Alex Balashov <[EMAIL PROTECTED]>wrote:
>
>> Yes.  There are some liabilities with that in that the signaling
>> messages may be incomplete (i.e. you may miss a BYE) and this is the
>> usual reason given for doing media proxying for more accurate accounting.
>>
>> But the latency, bandwidth consumption, and increased complexity and
>> cost associated with doing it on a large scale does not justify it, in
>> my opinion.  SIP-only accounting is "good enough" most of the time.
>>
>> Nuno Marques wrote:
>>
>> >
>> > Without mediaproxy? Only based on SIP messages?
>> >
>> >
>> >
>> > 2008/10/29 Alex Balashov <[EMAIL PROTECTED]
>> > >
>> >
>> > Nuno Marques wrote:
>> >
>> >  Every calls should pass through mediaproxy so that i can
>> > account them.
>> >
>> >
>> > You can do accounting without handling media.
>> >
>> > --
>> > Alex Balashov
>> > Evariste Systems
>> > Web: http://www.evaristesys.com/
>> > Tel: (+1) (678) 954-0670
>> > Direct : (+1) (678) 954-0671
>> > Mobile : (+1) (706) 338-8599
>> >
>> >
>>
>>
>> --
>> Alex Balashov
>> Evariste Systems
>> Web: http://www.evaristesys.com/
>> Tel: (+1) (678) 954-0670
>> Direct : (+1) (678) 954-0671
>> Mobile : (+1) (706) 338-8599
>>
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>
>
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>
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[asterisk-users] list-testing

[Johan Sandgren]

Had recent problems with the list, so checking I get list mails now :)

Sorry for the inconvenience,
Johan

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Mobile +46 70 173 4152
Box 1233, 221 05 Lund, Sweden
E-mail   [EMAIL PROTECTED]
Website www.svep.se

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Re: [asterisk-users] R key with Siemens Gigaset IP (was MWI with Siemens Gigaset S450IP)

[Olivier]

2008/10/29 Alan Lord <[EMAIL PROTECTED]>

> Olivier wrote:
> 
> > I'll reply to the correct thread
> >
> > [featuremap]
> > blindxfer => ## ; Blind transfer
> > ;disconnect => *0   ; Disconnect
> > ;automon => *1  ; One Touch Record
> > atxfer => A ; Attended transfer
> >
> > so set atxfer to 'A' and DTMF relay Application signal on the Gigaset
> to
> > 'A' (without quotes)
> >
> > and transfer works as expected
> >
> > Robb
> >
> > Thanks for replying !
> > I'll give it a try and report to the list
>
> I just tested this and it seems to work with the Siemens S685IPs. This
> thread was such a coincidence. We were trying to get attended transfer
> to work last night but setting the atxfer to "normal" things like *2
> just didn't work.
>
> I just set my S685IP base station to A for the Application Signal and
> set A in the features.conf and behold, when I now press the R key,
> Attended Transfer :-)
>
> Thanks
>
> Alan


Alan,

Did you get any success with MWI ?
With mine, Asterisk is getting 481 replies whenever Asterisk sends NOTIFY
updates.

Cheers


>
>
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[asterisk-users] list-testing2

[Johan Sandgren]

Had recent problems with the list, so checking I get list mails now :)

Sorry for the inconvenience,
Johan

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Phone +46 46 192 722
Mobile +46 70 173 4152
Box 1233, 221 05 Lund, Sweden
E-mail   [EMAIL PROTECTED]
Website www.svep.se

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