[asterisk-users] app_swift issues

[Yair Hakak]

Hi all,
 i'm trying to integrate cepstral and asterisk, and i have a problem i'd
appreciate any help with (i know it's a bit tangential, but i figure this is
the place with the most knowledge of app_swift and asterisk).
I've installed swift from cepstral.com with alison's voice, and it works
fine, from the command line i can do swift "hello there" -o test.wav and
then i play the wav and it includes the text. All good.
I've also installed app_swift according to the instructions here
http://www.mezzo.net/asterisk/app_swift.html, and "show application swift"
from the asterisk CLI brings up the application installed.

Now, when i put Swift('This is a test') in the extensions.conf file, i get
the following:
ERROR[3495]: app_cepstral.c:197 cepstral_speak: Failed to set voice.

I have not touched swift.conf (i'm using the defaults), and, i should add
that i have not yet purchased the cepstral voices so that when i run from
the command line it sticks "this voice is unlicensed..." or something like
that at the beginning of the file, if that makes some kind of difference.

I found the problem referenced here:
http://www.cepstral.com/forum/viewtopic.php?t=56&sid=baa6669e9958920393c62510caa47123&PHPSESSID=df1bcc629c4b8b37617d2d72c8b0232e
but
no solution...

any help will be most appreciated,

thanks,
 yair
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Re: [asterisk-users] Automatic provisioning of Sipura handsets (was: A linksys SPA921 behind NAT and firewall)

[Per Jessen]

Luki wrote:

> Here's how you do it.
> 
> 1) In the DHCP server's config (dhcpd.conf) you specify the IP of the
> TFTP server:
> option tftp-server-name "66.55.44.33";
> This can be a remote server, as long as it's accessible by the device.
> 
> 2) The factory settings on the Sipura devices (ATAs and phones) have
> /spa$PSN.cfg in the Provisioning profile rule, so the device will
> connect the TFTP server you specify and will try to retrieve that
> file, i.e. ftfp://66.55.44.33/spa942.cfg for the SPA-942 in this
> example.
> 
> 3) This file contains very minimal information, which tells the device
> where to download its final configuration from. This can be a remote
> http server so you can maintain the configs on one central server.
> Example:
> 
> 
>  
>http://YOUR.HTTP.PROVISIONING.SERVER.HOST/$MA.bin
>  
> 

Is it possible that this only works with a "compiled" config?  I've been
trying do the above, but with the XML config, and I'm not getting
anywhere. 

> 4) The device will then connect via HTTP and will try to retrieve for
> configuration for its MAC address. Since it's a HTTP request, you can
> generate the provisioning data on the fly (even from the a database),
> either in XML format or in compiled format if you have the Sipura
> compiler.

Oh well - I wonder what I'm doing wrong then.  I've been trying to get
this to work for most of last week. 

> The above works just fine and very reliably. We have disabled periodic
> resync as the Sipura phones seem to reboot sometimes for no good
> reason when they apply the "new" but unchanged profile. If there is a
> config change, we just push it on the phone with SIP NOTIFY option.

Do you push it from Asterisk or somewhereelse?  Again, I can't make it
work. I've got "Auth resync-reboot" disabled on the SPA, so it
shouldn't be asking for authentication, but the SIP NOTIFY goes out,
and the phone does nothing.


/Per Jessen, Zürich

-- 
http://www.spamchek.com/ - your spam is our business.


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Re: [asterisk-users] cmd mysql

[Benchev]

> I've been using mysql databases more and more.  I've run
> across a couple
> of instances where I've either made a mistake on the ip
> address of the
> mysql database or for whatever reason, mysql wasn't
> running.  In those
> instances, I've noted that the mysql command will hang
> indefinitely
> (I've counted to 40 before killing it).
>
> The offending line is:
>
> exten => s,1,MYSQL(Connect connid 192.168.103.15
> mysqladmin 'x' did)
>
> Is there a way to specify a timeout for the mysql
> command?
>
> I'm not finding anything on Google, voip-info or the
> documents in the
> add-ons directory.  Any help would be appreciated.
>
By experience, after every query sequence you should use
MYSQL(Clear ${resultid}) otherwise they'll become 400.
As it is said into app_addon_sql_mysql.c
"Frees memory and datastructures associated with result
set."

Hope that helps,
Benchev

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Re: [asterisk-users] Receptionists Phone suggestions? (Not Snom370)

[Paul Hales]

Agreed - handling multiple calls and transferring them on a Snom is a
problem. Too fiddly.

Polycom phones work well in reception situations, if set up well.

Haven't tested the new Aastra's (but the Aastra transfer function works
well) but they would probably be OK too.

PaulH


On Fri, 2007-10-19 at 09:12 +0100, Russell Brown wrote:
> Does anyone have any suggestions for a decent receptionists phone?
> Aastra?  Grandstream?
> 
> Something with (potentially) lots of BLFs, large(ish) screen, headset
> and most importantly the ability to transfer calls?
> 
> I've installed five Snom 370s that seemed ideal but my client is very
> very unhappy as the Snom 370 can't transfer a call correctly if there's
> another call coming in (details below if you/re interested).  I've
> verified this problem with Snom who's response is that the receptionist
> should answer all of the incoming calls before trying to do a transfer -
> 
> That's just Bonkers!
> 
> So... any suggestions?
> 
> 
> Details of Snom 370 problem for the record:
> 
> Snom370 gets a Call (Call A). 
> Snom370 answers Call A. Call A wants to be transferred to Phone C. 
> Snom370 has another call ringing (Call B). 
> Snom370 presses HOLD button gets Dialtone. Call A is on Hold, Call B
> still ringing. 
> Snom370 Dials Phone C (Call C). 
> Snom370 talks to Call C. 
> Snom370 presses TRANSFER. 
>  
> The display shows: 
>   
> < CallA 
> > CallB 
> 
> The soft keys now show "<<" and ">>". Pressing them does nothing. 
> 
> When the TRANSFER button is pressed again, CallA is connected to CallB
> (the original caller is now talking to the previously unanswered party)
> not what one wanted to happen!
> 
> It's not difficult to see why my client is throwing their toys out of
> the pram and I'm going to have to replace the Snoms at my expense :-(
> 
> 


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[asterisk-users] Issues with Zaptel 1.4.5.1

[Aldo D. Sudak]

Hi Tzafrir!

Thank you for your answer, and my apologies for my delayed 
response. I regret to say that the patch test's results were not succesful. 
I shall describe the whole procedure in detail for you to establish whether
I did something wrong. 

The fact is that I am as much a Linux beginner as I am an Asterisk one, 
and so I had first to learn how to stick the patch. Since I found no 
instructions to apply it, perhaps this was not the right procedure... After 
initially attempting with 3M self-adhessive tape with little success..., I 
finally 
did the following:

1) I copied the autoztcgf.diff to the /usr/src/zaptel-1.4.5.1 directory
2) I run patch -b < autoztcfg.diff
3) Since a file named zaptel.c did not exist, patch offered me to enter the 
filename. I entered zaptel-base.c which I thought was the most probable 
file to be patched.
4) I obtained 6 out of 6 success messages indicating however very variable 
offsets, ranging between 13 and 224 lines, which called my attention. May 
be this was normal, however.

Then I performed the following actions:

1) I sent the zaptel file in /etc/modprobe.d/ to trash (remember this file had 
been artificially grafted by me by means of zaptel 1.4.4). By the way I want 
to make clear that zaptel files in /etc/init.d and in /etc/rc.d/init.d were 
created 
by version 1.4.5.1, and that zaptel modules were loaded during bootup (they 
just were not configured).
2) I stopped Asterisk and unloaded zttranscode, wctdm24xxp and zaptel by 
means of rmmod. (My card is a TDM800P)
3) I then installed the patched zaptel-1.4.5.1 starting with a make clear 
command. 
Everything went OK. A new zaptel.conf replaced the previous one. I properly 
configured it by hand. I restarted the machine.

Results:

1) Zaptel was not automatically configured. I ran zttool: 'unconfigured' 
statement
2) I ran /sbin/ztcfg -vv. The result was a wonderful computer crash. I had to 
restart 
it from the CPU button :-(

Finally I returned to my original installation and re-sent the zaptel script to 
the 
/etc/modprobe.d directory. Everything began to work properly again. In my 
humble and newbie's opinion, a patch to simply install this script would have 
to 
solve the problem. Considering that previous versions did it, I cannot 
understand 
well why this action has been eliminated in version 1.4.5.1.

Greetings,

Aldo

---

On Wed Oct 17 16:57:31 CDT 2007 Tzafrir Cohen wrote:

>On Wed, Oct 17, 2007 at 06:37:21PM -0300, Aldo D. Sudak wrote:
>> Greetings to all list members!
>> 
>> My name is Aldo Sudak and I am an Asterisk newbie. I am writing now because 
>> I have not 
>> been able to find any mention to the issues described below, neither in this 
>> list nor in the wiki.
>> 
>> I am performing preliminary tests with Asterisk 1.4.11 and Zaptel 1.4.5.1 on 
>> Fedora Core 6. 
>> Installation of all of the Asterisk packages was straightforward and I 
>> always obtained success 
>> messages. Following Asterisk and Zaptel installation I ran the corresponding 
>> make config 
>> commands in order to have both Asterisk and Zaptel running at startup. With 
>> Zaptel, however, 
>> two issues arose:
>> 
>> 1) zaptel.conf in the /etc/ directory had not been created. The GUI loaded 
>> itself a very rudimentary 
>> zaptel.conf file which lacked loadzone and defaultzone definitions, so I 
>> prefered to manually copy 
>> and paste zaptel.conf.sample from the sources and rename it.
>
>use zapconf or genzaptelconf to generate one that actually works.
>Both are included with Zaptel and installed by default (genzaptelconf is
>being phased out).
>
>> 
>> 2) Zaptel modules loaded during bootup but were not automatically 
>> configured, and so I had to 
>> manually run ztcfg and afterwards restart Asterisk each time I started the 
>> computer. I discovered 
>> that there did not exist a 'zaptel' file in the /etc/modprobe.c/ directory, 
>> so I made an experiment 
>> by installing Zaptel 1.4.4, which generated the file, and then re-installing 
>> Zaptel 1.4.5.1. The 
>> experience was successful and now Zaptel 1.4.5.1 is loaded and configured 
>> during bootup.
>
>  /etc/init.d/zaptel start
>
>is required to properly start zaptel. It will also run ztcfg. Otherwise
>users run into issues where misconfigured zaptel.conf fails loading of a
>module. That is a buggy behaviour.
>
>If your card is an analog one, take a look at http://bugs.digium.com/7613
>and tell me what you think.
>
>Something similar for digital spans would require more information in
>sysfs.
>
>-- 
>   Tzafrir Cohen   
>icq#16849755  jabber:tzafrir.cohen at xorcom.com
>+972-50-7952406   mailto:tzafrir.cohen at xorcom.com   
>http://www.xorcom.com  iax:guest at local.xorcom.com/tzafrir

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[asterisk-users] Asterisk Initial Set-up - 'Registration Refused' at FWD

[Aldo D. Sudak]

Hi to all!

I'm glad to be of any help. I had the same issue and sent an e-mail directly 
to FWD. I paste the answer below.

Greetings, 

Aldo

- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - 
- - - - - - - - - - - - - - - - - - - - - -


Aldo,

FWD IAX is experimental and not supported. Please create a SIP trunk
which will be more reliable.

Regards,

Juan Vides


On 10/16/07, Aldo D. Sudak <[EMAIL PROTECTED]> wrote:
>
>
> Dear sirs,
>
> I am aware that this is perhaps not the appropriate address to write to, but
> I was not able to find any solution on the Internet. So I shall be very
> grateful if you send me an answer suggesting the possible cause for my
> problem.
>
> I have recently obtained an account at FWD, #869178. I have enabled IAX
> service for this account. I then configured my Asterisk server according to
> your instructions but always obtain a 'registration rejected' message. In
> order to verify whether this was an Asterisk configuration problem, I
> introduced my account data to an Iax-Lite soft phone installed in another
> computer but the result was the same. On the other hand, I configured an
> X-Lite SIP phone for this account and it worked fine. Same number, same
> password; so I wonder if there is something happening related with your
> iax2.fwdnet.net server.
>
> Thank you in advance.
>
>
> Aldo D. Sudak


-- 
Juan Vides

Fax:631-293-3996
 115 Broadhollow Road, Suite 225
Melville, NY 11747
www.freeworlddialup.com

 FWD#  393464
Yahoo: pulverCommunicator
MSN:[EMAIL PROTECTED]
AOL:plvrcomcator
 ICQ: 341693563

El mail recibido, fue verificado por el Servicio de Antivirus para mail de 
Fullzero
y se encuentra libre de virus

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Re: [asterisk-users] cmd mysql

[Doug Lytle]

Baji Panchumarti wrote:
> you may be confusing the db server because mysqladmin

I made this name up, it's not the user name that I really use.


>
>  no quotes around the password.
>
I'll test this out, thanks.

Doug


-- 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."



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Re: [asterisk-users] cmd mysql

[Baji Panchumarti]

  On 10/21/07, Doug Lytle  wrote:

> Hey everybody,
>
> I've been using mysql databases more and more.  I've run across a couple
> of instances where I've either made a mistake on the ip address of the
> mysql database or for whatever reason, mysql wasn't running.  In those
> instances, I've noted that the mysql command will hang indefinitely
> (I've counted to 40 before killing it).
>
> The offending line is:
>
> exten => s,1,MYSQL(Connect connid 192.168.103.15 mysqladmin 'x'
did)

 there are a couple of things wrong with your MYSQL() call,
 you may be confusing the db server because mysqladmin
 the name of a mysql util, pick another username.

 no quotes around the password.

 Here are my statements, and they don't hang if mysql isn't
 running on the db server, the * console shows a connect
 failed error and continues :

 exten => s,n,Set(db_host=192.168.111.11)
 exten => s,n,MYSQL(Connect db_connid ${db_host} user pass dbname)

> Is there a way to specify a timeout for the mysql command?
>
> I'm not finding anything on Google, voip-info or the documents in the
> add-ons directory.  Any help would be appreciated.

more info here :

http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MYSQL

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[asterisk-users] Hanging up all call on a device via CLI/AMI/AGI

[Forrest Vodden]

Hello, never posted to a mailing list before. I've been trying to work out this 
problem for quite awhile now. I have a PHP script which is run whenever an 
emergency situation happens. The script connects to the AMI and originates 
calls to previously defined "emergency" extensions. I'm looking for a way to 
disconnect all calls on a device if it is in use, in order to deliver the 
message. I cannot use SoftHangup with the 'a' option within the dial plan, 
because if the device that is called is busy, then the context is never hit. It 
doesn't appear that their is a way within the AMI or using AGI or even within 
the CLI to disconnect all calls on a device, only is you know the entire 
channel name.

For instance, from the CLI, running 'soft hangup SIP/250-ab627038' will 
disconnect the channel successfully. But the problem is, I have no way of 
finding out the full channel name from within my my script using the AMI. The 
only possibility I see is running the "show channels" CLI command via the AMI 
Command action, grabbing the resulting list and parsing the Channel Name column 
before the script begins calling extensions. Then before each extension is 
called, check to see if the extension exists in the retrieved channel list. If 
so, disconnect it.

Anyways, it seems odd to me that the 'a' option is available from within the 
dial plan, but not anywhere else. That would be the simplest approach to this.

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Re: [asterisk-users] Automatic provisioning of Sipura handsets (was: A linksys SPA921 behind NAT and firewall)

[Luki]

> I'd like to be able to templatize a server, add a bunch of new handsets
> into sip.conf and extensions.conf, and then plug the phones into a
> network and have some DHCP and/or TFTP "glue" logic that sees the DHCP
> or TFTP request, and from it generates a boot file (an .XML file) and a
> response parameter list for DHCP... populates a file into the /tftpboot/
> directory, etc.

Here's how you do it.

1) In the DHCP server's config (dhcpd.conf) you specify the IP of the
TFTP server:
option tftp-server-name "66.55.44.33";
This can be a remote server, as long as it's accessible by the device.

2) The factory settings on the Sipura devices (ATAs and phones) have
/spa$PSN.cfg in the Provisioning profile rule, so the device will
connect the TFTP server you specify and will try to retrieve that
file, i.e. ftfp://66.55.44.33/spa942.cfg for the SPA-942 in this
example.

3) This file contains very minimal information, which tells the device
where to download its final configuration from. This can be a remote
http server so you can maintain the configs on one central server.
Example:


 
   http://YOUR.HTTP.PROVISIONING.SERVER.HOST/$MA.bin
 


4) The device will then connect via HTTP and will try to retrieve for
configuration for its MAC address. Since it's a HTTP request, you can
generate the provisioning data on the fly (even from the a database),
either in XML format or in compiled format if you have the Sipura
compiler.

The above works just fine and very reliably. We have disabled periodic
resync as the Sipura phones seem to reboot sometimes for no good
reason when they apply the "new" but unchanged profile. If there is a
config change, we just push it on the phone with SIP NOTIFY option.

--Luki

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Re: [asterisk-users] Prompting for number when CID number not sent?

[Vincent]

On Sun, 21 Oct 2007 13:31:28 -0400, Doug Lytle <[EMAIL PROTECTED]>
wrote:
>You'll want to look at the Privacy Manager:

Great :-) I'll take a look... once I can get the TDM card to pass the
CID number to Asterisk when it's actually sent by the telco.

Thanks for the tip.


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Re: [asterisk-users] Asterisk Initial Set-up - 'Registration Refused' at FWD

[Alan Lord]

Ian Hodgson wrote:
> Hello,
> Sorry for what may be a basic question, but I have spent a number of 
> hours trawling the forums without resolving the problem, and hence this 
> post.
 >
> I have just started to dabble with Asterisk, as much for the 
> learning than anything else. I created an account on FWD and used the 
> Asterisk settings that the FWD web site recommends at 
> http://www.freeworlddialup.com/help/?p=knowledgebase&c=18&a=76 
> .
>  
> I started the Asterisk server, and after the set of start-up 
> messages, I see...
>  
> Oct 21 16:34:53 NOTICE[9444]: chan_iax2.c:7597 socket_read: Registration 
> of '867245' rejected: 'Registration Refused' from: '192.246.69.186'

I am pretty sure this a problem with FWD. If you read their forums on 
IAX, the moderators repeatedly state that IAX support is experimental 
and not well supported.

I have exactly the same issue as you do...

> From the Google searches that I have done, I cannot find a solution 
> to this. I have tried opening up UDP port 4569 on my Netgear router and 
> setting port forwarding to my Asterisk PC, but this makes no difference. 
> I am sure that I have missed something simple, so please do not worry 
> about stating what may seem to be obvious. Thanks in advance for your help.
>  
> Ian.

Nope - I haven't had to change the firewall on my router and I 
successfully have an IAX-IAX trunk running between my Asterisk server 
and my business partners. I think it's an FWD problem.

Alan

-- 
The way out is open!
http://www.theopensourcerer.com


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Re: [asterisk-users] Extensions.conf for basic IVR?

[Alan Lord]

Vincent wrote:
> On Sat, 20 Oct 2007 11:37:56 +0100, Alan Lord <[EMAIL PROTECTED]>
> wrote:
>> Look back a few hours in this mailing list for the message called " 
>> IAX2: Incoming calls answered prematurely[RESOLVED]".
>>
>> I have included most of how I setup a simple IVR. It wasn't that hard to 
>> do and I have only been using asterisk for a week or so...
> 
> Thanks for the tip.
> 
> http://lists.digium.com/pipermail/asterisk-users/2007-October/198815.html
> 

Hi, make sure you read the post with [RESOLVED] in the subject line. It 
is updated and "working" [for me anyway].

http://lists.digium.com/pipermail/asterisk-users/2007-October/198857.html

Cheers

Alan

-- 
The way out is open!
http://www.theopensourcerer.com


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[asterisk-users] cmd mysql

[Doug Lytle]

Hey everybody,

I've been using mysql databases more and more.  I've run across a couple
of instances where I've either made a mistake on the ip address of the
mysql database or for whatever reason, mysql wasn't running.  In those
instances, I've noted that the mysql command will hang indefinitely
(I've counted to 40 before killing it).

The offending line is:

exten => s,1,MYSQL(Connect connid 192.168.103.15 mysqladmin 'x' did)

Is there a way to specify a timeout for the mysql command?

I'm not finding anything on Google, voip-info or the documents in the
add-ons directory.  Any help would be appreciated.

Thanks,

Doug



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Re: [asterisk-users] WARNING: chan_zap.c process_zap: Ignoring signalling

[Brian West]

Thats a great step forward.  Auto for PRI doesn't make sense... but  
two configs to describe the same thing makes no sense.

/b

On Oct 21, 2007, at 1:03 PM, Tzafrir Cohen wrote:

> On Sun, Oct 21, 2007 at 11:57:45AM -0500, Brian West wrote:
>> It actually CAN but because someone was lazy and didn't want to
>> actually do the work to make it possible to do a full change during a
>> reload.  The biggest issue is ztcfg would have to be absorbed into
>> chan_zap to make it 100% possible.  In fact if Digium wanted to make
>> Asterisk easier to configure/setup they would merge ztcfg into
>> chan_zap and get rid of /etc/zaptel.conf and save a config step.
>
> Look at the "auto" signalling in
> http://svn.digium.com/svn/asterisk/team/group/zapata_conf .
>
> This is rather nice for analog channels. Much less of a help, I'm
> afraid, for PRI.
>
> -- 
>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] WARNING: chan_zap.c process_zap: Ignoring signalling

[Tzafrir Cohen]

On Sun, Oct 21, 2007 at 11:57:45AM -0500, Brian West wrote:
> It actually CAN but because someone was lazy and didn't want to  
> actually do the work to make it possible to do a full change during a  
> reload.  The biggest issue is ztcfg would have to be absorbed into  
> chan_zap to make it 100% possible.  In fact if Digium wanted to make  
> Asterisk easier to configure/setup they would merge ztcfg into  
> chan_zap and get rid of /etc/zaptel.conf and save a config step.

Look at the "auto" signalling in
http://svn.digium.com/svn/asterisk/team/group/zapata_conf .

This is rather nice for analog channels. Much less of a help, I'm
afraid, for PRI.

-- 
   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] Prompting for number when CID number not sent?

[Doug Lytle]

Vincent wrote:
> How would I go about prompting users for their phone number?
>
>   

You'll want to look at the Privacy Manager:

http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+PrivacyManager

Doug


-- 
Ben Franklin quote:

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Safety, deserve neither Liberty nor Safety."



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Re: [asterisk-users] WARNING: chan_zap.c process_zap: Ignoring signalling

[Brian West]

It actually CAN but because someone was lazy and didn't want to  
actually do the work to make it possible to do a full change during a  
reload.  The biggest issue is ztcfg would have to be absorbed into  
chan_zap to make it 100% possible.  In fact if Digium wanted to make  
Asterisk easier to configure/setup they would merge ztcfg into  
chan_zap and get rid of /etc/zaptel.conf and save a config step.

/b

On Oct 21, 2007, at 10:23 AM, Tzafrir Cohen wrote:

> On Sun, Oct 21, 2007 at 04:27:17PM +0200, Vincent wrote:
>
>> ubuntu*CLI> reload chan_zap.so
>> -- Reloading module 'chan_zap.so' (Zapata Telephony)
>>   == Parsing '/etc/asterisk/zapata.conf': Found
>> [Oct 21 16:22:37] WARNING[8240]: chan_zap.c:11120 process_zap:
>> Ignoring signalling
>
> chan_zap cannot change signalling of a channel on reload. So that
> parameter is ignored on reload.
>
> False warning...
>
> -- 
>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] Automatic provisioning of Sipura handsets (was: A linksys SPA921 behind NAT and firewall)

[Anthony Francis]

>
> Anselm Martin Hoffmeister wrote:
>   
>> The problem there is that you have a very small "windows". AFAIK there
>> are no tftp servers that can generate files on-the-fly, so your script
>>
>> 
>   

You could make a perl script that pretends to be a TFTP server. Then it 
could generate the file on the fly.

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[asterisk-users] Asterisk Initial Set-up - 'Registration Refused' at FWD

[Ian Hodgson]

Hello,
Sorry for what may be a basic question, but I have spent a number of
hours trawling the forums without resolving the problem, and hence this
post.

I have just started to dabble with Asterisk, as much for the learning
than anything else. I created an account on FWD and used the Asterisk
settings that the FWD web site recommends at
http://www.freeworlddialup.com/help/?p=knowledgebase&c=18&a=76.

I started the Asterisk server, and after the set of start-up messages, I
see...

Oct 21 16:34:53 NOTICE[9444]: chan_iax2.c:7597 socket_read: Registration of
'867245' rejected: 'Registration Refused' from: '192.246.69.186'

From the Google searches that I have done, I cannot find a solution to
this. I have tried opening up UDP port 4569 on my Netgear router and setting
port forwarding to my Asterisk PC, but this makes no difference. I am sure
that I have missed something simple, so please do not worry about stating
what may seem to be obvious. Thanks in advance for your help.

Ian.
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Re: [asterisk-users] WARNING: chan_zap.c process_zap: Ignoring signalling

[Vincent]

On Sun, 21 Oct 2007 17:23:03 +0200, Tzafrir Cohen
<[EMAIL PROTECTED]> wrote:
>chan_zap cannot change signalling of a channel on reload. So that
>parameter is ignored on reload.
>
>False warning...

OK. So to check that Zaptel is correctly configured, I can just type
"zap show channels" in the CLI.

Thanks.


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Re: [asterisk-users] WARNING: chan_zap.c process_zap: Ignoring signalling

[Vincent]

On Sun, 21 Oct 2007 10:59:49 -0400, "C F" <[EMAIL PROTECTED]> wrote:
>I believe that by reloading without restarting asterisk doesnt reload
>the signalling part

Thanks for the help. I did read this somewhere, so I typed "stop now"
in the CLI, followed by "safe_asterisk", "asterisk -r", and "reload":
I still get the warning:


  == Parsing '/etc/asterisk/zapata.conf': Found
[Oct 21 17:23:48] WARNING[8419]: chan_zap.c:11120 process_zap:
Ignoring signalling
-- Reconfigured channel 1, FXS Kewlstart signalling


I have no idea where to look, since the two configuration files look
like those I read on the Net.


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[asterisk-users] Prompting for number when CID number not sent?

[Vincent]

Hi

The first step I have to go through when users call into our
IVR is to handle the case where users' PBX hides their CID number. In
that case, I need to have them type their phone number (ten digits).

OTOH, those who call without hiding their CID number are sent directly
to the main menu.

How would I go about prompting users for their phone number?

Here's what I have at this point:

exten => s,1,Answer

;CID sent : go to main_menu
exten => s,n,Background(/root/asterisk_sound_files/main_menu)
;CID hidden : prompt for CID and loop until ok
exten => s/,n,Playback(/root/asterisk_sound_files/no_cid)

exten =>
_[1-4],1,Playback(/root/asterisk_sound_files/you_can_record_msg)
exten => _[1-4],n,Record(/tmp/asterisk-msg:wav)
exten => _[1-4],n,Wait(1)
exten => _[1-4],n,Playback(/tmp/asterisk-msg)
exten => _[1-4],n,wait(1)

;here, rewrite CID name by looking up CID # in database
;put CID name + number in variables
;exten => _[1-4],n,SetVar(cid=${callerid})
;send e-mail with CID name + number and link to WAV file to people in
charge of selected software

exten => _[1-4],n,Hangup


Thanks for any tip.


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Re: [asterisk-users] WARNING: chan_zap.c process_zap: Ignoring signalling

[Tzafrir Cohen]

On Sun, Oct 21, 2007 at 04:27:17PM +0200, Vincent wrote:

> ubuntu*CLI> reload chan_zap.so
> -- Reloading module 'chan_zap.so' (Zapata Telephony)
>   == Parsing '/etc/asterisk/zapata.conf': Found
> [Oct 21 16:22:37] WARNING[8240]: chan_zap.c:11120 process_zap:
> Ignoring signalling

chan_zap cannot change signalling of a channel on reload. So that
parameter is ignored on reload.

False warning...

-- 
   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] WARNING: chan_zap.c process_zap: Ignoring signalling

[C F]

I believe that by reloading without restarting asterisk doesnt reload
the signalling part

On 10/21/07, Vincent <[EMAIL PROTECTED]> wrote:
> Hello
>
>   I've been googling for this message, but can't find why
> Asterisk sends a warning. The configuration files look similar to
> http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf.sample
>
> It's a TDM card with just one FXO module on it, and I connected an
> RJ11 cable to it from the wall plug:
>
> # cat /etc/zaptel.conf
> fxsks=1
> loadzone=fr
> defaultzone=fr
>
> # ztcfg -vv
> Zaptel Version: 1.4.5.1
> Echo Canceller: MG2
> Configuration
> ==
>
>
> Channel map:
>
> Channel 01: FXS Kewlstart (Default) (Slaves: 01)
>
> 1 channels configured.
>
> # lsmod | grep zap
> zaptel189860  7 zttranscode,ztdummy,wctdm
> crc_ccitt   3072  2 zaptel,hisax
>
> # cat /etc/asterisk/zapata.conf
> [channels]
> context=my-phones
> usecallerid=yes
> hidecallerid=no
> immediate=no
>
> signalling=fxs_ks
> echocancel=yes
> channel => 1
>
> ubuntu*CLI> reload chan_zap.so
> -- Reloading module 'chan_zap.so' (Zapata Telephony)
>   == Parsing '/etc/asterisk/zapata.conf': Found
> [Oct 21 16:22:37] WARNING[8240]: chan_zap.c:11120 process_zap:
> Ignoring signalling
> -- Reconfigured channel 1, FXS Kewlstart signalling
>   == Parsing '/etc/asterisk/users.conf': Found
> ubuntu*CLI>
>
> Any idea? Should I just ignore this warning?
>
> Thank you.
>
>
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[asterisk-users] WARNING: chan_zap.c process_zap: Ignoring signalling

[Vincent]

Hello

I've been googling for this message, but can't find why
Asterisk sends a warning. The configuration files look similar to
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf.sample

It's a TDM card with just one FXO module on it, and I connected an
RJ11 cable to it from the wall plug:

# cat /etc/zaptel.conf 
fxsks=1
loadzone=fr
defaultzone=fr

# ztcfg -vv
Zaptel Version: 1.4.5.1
Echo Canceller: MG2
Configuration
==


Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)

1 channels configured.

# lsmod | grep zap
zaptel189860  7 zttranscode,ztdummy,wctdm
crc_ccitt   3072  2 zaptel,hisax

# cat /etc/asterisk/zapata.conf 
[channels]
context=my-phones
usecallerid=yes
hidecallerid=no
immediate=no

signalling=fxs_ks
echocancel=yes
channel => 1

ubuntu*CLI> reload chan_zap.so
-- Reloading module 'chan_zap.so' (Zapata Telephony)
  == Parsing '/etc/asterisk/zapata.conf': Found
[Oct 21 16:22:37] WARNING[8240]: chan_zap.c:11120 process_zap:
Ignoring signalling
-- Reconfigured channel 1, FXS Kewlstart signalling
  == Parsing '/etc/asterisk/users.conf': Found
ubuntu*CLI> 

Any idea? Should I just ignore this warning?

Thank you.


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[asterisk-users] Questions about Plycom phones and Asterisk

[Yehavi Bourvine +972-8-9489444]

Hello,

  I have two issues which I would like to know whether someone has an answer to
them:

1. Our institute has over 8,000 phone numbers and I would like to allow
   people to search it from the phone. I am willing to write some XHTML
   scripts to run through the microbrowser, but I cannot find at the
   documentation what format it should be in order for the phone to understand
   the output and allow dialling of the numbers returned.

2. Polycom supports external conference server via the parameter
   voIpProt.SIP.conference.address. Any idea how should I include in it the
   conference room number (so I can pass it to MeetMe() application)?

 Thanks! __Yehavi:

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Re: [asterisk-users] IMAP usage with Asterisk

[Yehavi Bourvine +972-8-9489444]

> Yehavi Bourvine +972-8-9489444 wrote:
>>   In any case, I'll try this week to upgrade to 1.4.6 version and then add 
>> IMAP
>> support and inform what happens.
>
> There have been _many_ IMAP related fixes sine 1.4.6.  Please try the latest
> version, 1.4.13, instead.
>
> --
> Russell Bryant

Sorry, I had a typing mistake - I meant 1.4.13...

Anyway, I tried it at the lab today and it works ok. There is one minor problem
when parsing the IMAP headers (files a bug at ID 11043).

I'll be able to move it to the production system only next week.

 Thanks, __Yehavi:

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Re: [asterisk-users] Automatic provisioning of Sipura handsets (was: A linksys SPA921 behind NAT and firewall)

[Per Jessen]

Anselm Martin Hoffmeister wrote:

> Am Samstag, den 20.10.2007, 22:58 -0700 schrieb Philip Prindeville:
>> I'd like to be able to templatize a server, add a bunch of new
>> handsets into sip.conf and extensions.conf, and then plug the phones
>> into a network and have some DHCP and/or TFTP "glue" logic that sees
>> the DHCP or TFTP request, and from it generates a boot file (an .XML
>> file) and a response parameter list for DHCP... populates a file into
>> the /tftpboot/ directory, etc.
>> 
>> How viable is this?
> 
> The problem there is that you have a very small "windows". AFAIK there
> are no tftp servers that can generate files on-the-fly, so your script
> would have to generate the XML within less than a second, reliably,
> and do all the necessary asterisk changes within another second or
> two, and I doubt this will be possible _that_ quick.

Perhaps you could trigger the creation of the config, xml etc. on the
first TFTP request - on the retry the files would then be ready to go.

> Of course you can use ISC dhcpd for tailoring answers to your needs
> (dynamic setting of config file etc), but IMO this will only work well
> if the phones support http config download, 

The SPA-9x1 does support http download, but I don't see how you could
change the initial TFTP request to HTTP without manually configuring
the phone.  Even then I'm not sure it would work - I certainly haven't
managed to make any of my SPAs do an auto-config over HTTP. 


/Per Jessen, Zürich

-- 
http://www.spamchek.com/ - your spam is our business.


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Re: [asterisk-users] Sometimes echoes & Asterisk sometimes connects tooearly

[Michael J. Liberatore]

I have had ongoing echo problems with snom 360's, maybe the problem lies
with your phones...

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stefan
Guenther
Sent: Sunday, October 21, 2007 5:20 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Sometimes echoes & Asterisk sometimes connects
tooearly

Hello,

I have read the articles on echo cancellation
(http://www.voip-info.org/wiki/view/Asterisk), but couldn't find a
solution to my problem.

We are running Asterisk 1.4.12 together with an EICON DIVA SERVER BRI-2M
PCI (current driver from EICON) and some SNOM 300/360.
There are few clients where we recognize echoes on both sides when we
call them via ISDN.
With some of these clients we don't even hear their name, when they pick
up the phone, because Asterisk connects the call to early.
I don't know whether these two effects belong together but both are
rather disturbing.

Here is our capi.conf:

[general]
nationalprefix=0
internationalprefix=00
rxgain=1
txgain=0.8
language=de

[ISDN1]
incomingmsn=8304498,8304499
isdnmode=msn
group=1
controller=1
softdtmf=1
context=isdnin
echosquelch=2
echocancel=yes
echotail=64
callgroup=1
devices=2

I have changed the values for rxgain and txgain but that didn't change
much.

Thanks for any hint or advice,

Stefan

-- 


in-put GbR - Das Linux-Systemhaus
Stefan-Michael Guenther
Geschaeftsfuehrer
Moltkestrasse 49 D-76133 Karlsruhe
Tel./Fax : +49 (0)721 / 83044 - 98/93
http://www.in-put.de

 Schulungen  Installationen
 Beratung   Support
  Voice-over-IP-Loesungen




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[asterisk-users] Sometimes echoes & Asterisk sometimes connects too early

[Stefan Guenther]

Hello,

I have read the articles on echo cancellation
(http://www.voip-info.org/wiki/view/Asterisk), but couldn't find a
solution to my problem.

We are running Asterisk 1.4.12 together with an EICON DIVA SERVER BRI-2M
PCI (current driver from EICON) and some SNOM 300/360.
There are few clients where we recognize echoes on both sides when we
call them via ISDN.
With some of these clients we don't even hear their name, when they pick
up the phone, because Asterisk connects the call to early.
I don't know whether these two effects belong together but both are
rather disturbing.

Here is our capi.conf:

[general]
nationalprefix=0
internationalprefix=00
rxgain=1
txgain=0.8
language=de

[ISDN1]
incomingmsn=8304498,8304499
isdnmode=msn
group=1
controller=1
softdtmf=1
context=isdnin
echosquelch=2
echocancel=yes
echotail=64
callgroup=1
devices=2

I have changed the values for rxgain and txgain but that didn't change much.

Thanks for any hint or advice,

Stefan

-- 


in-put GbR - Das Linux-Systemhaus
Stefan-Michael Guenther
Geschaeftsfuehrer
Moltkestrasse 49 D-76133 Karlsruhe
Tel./Fax : +49 (0)721 / 83044 - 98/93
http://www.in-put.de

 Schulungen  Installationen
 Beratung   Support
  Voice-over-IP-Loesungen




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Re: [asterisk-users] Extensions.conf for basic IVR?

[Vincent]

On Sat, 20 Oct 2007 11:37:56 +0100, Alan Lord <[EMAIL PROTECTED]>
wrote:
>Look back a few hours in this mailing list for the message called " 
>IAX2: Incoming calls answered prematurely  [RESOLVED]".
>
>I have included most of how I setup a simple IVR. It wasn't that hard to 
>do and I have only been using asterisk for a week or so...

Thanks for the tip.

http://lists.digium.com/pipermail/asterisk-users/2007-October/198815.html


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Re: [asterisk-users] Automatic provisioning of Sipura handsets (was: A linksys SPA921 behind NAT and firewall)

[Anselm Martin Hoffmeister]

Am Samstag, den 20.10.2007, 22:58 -0700 schrieb Philip Prindeville:
> Erik Anderson wrote:
> > On 10/20/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> >   
> >> If you are trying to use non-complied ("XML") profiles... don't even
> >> bother wasting your time.
> >> 
> >
> > Why is that?  I'm using the xml-style config and they're working just fine.
> >
> >   
> 
> I'd like to be able to templatize a server, add a bunch of new handsets 
> into sip.conf and extensions.conf, and then plug the phones into a 
> network and have some DHCP and/or TFTP "glue" logic that sees the DHCP 
> or TFTP request, and from it generates a boot file (an .XML file) and a 
> response parameter list for DHCP... populates a file into the /tftpboot/ 
> directory, etc.
> 
> How viable is this?

The problem there is that you have a very small "windows". AFAIK there
are no tftp servers that can generate files on-the-fly, so your script
would have to generate the XML within less than a second, reliably, and
do all the necessary asterisk changes within another second or two, and
I doubt this will be possible _that_ quick.

Of course you can use ISC dhcpd for tailoring answers to your needs
(dynamic setting of config file etc), but IMO this will only work well
if the phones support http config download, because that gives you a
much better hook to put your script, and you can hold back the file
until all the asterisk changes are done, and finally return the XML (or
whatever).

BR
Anselm


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Re: [asterisk-users] A linksys SPA921 behind NAT and firewall

[Per Jessen]

[EMAIL PROTECTED] wrote:

> If you are trying to use non-complied ("XML") profiles... don't even
> bother wasting your time.

Oh.  I _am_ using the XML format.  When I initiate a resync over the
http server, it works fine, except the SPA doesn't start the regular
resync.



/Per Jessen, Zürich

-- 
http://www.spamchek.com/ - your spam is our business.


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Re: [asterisk-users] SIP Peer with Mulit Host

[Abdul]

This is little risky, 
if some one got his account username/pws he will be able to send the traffic
allowing only IPs means he need to assign his IP then he can send traffic.

Is there no possibilities in asterisk to adding more host?

Thank You








> If you are only going to receive and 
>not send calls to that host then use

> host=dynamic

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[asterisk-users] Asterisk H323 Config

[Arun Kumar]

Hi

Need help on this setup:

Incoming DID in H323  > Asterisk Server --> SIP Phone


please tell me to achieve this above setup what needs to be done in
Asterisk.


thanks

Arun
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