RE: [Asterisk-Users] ser <-> asterisk configs anyone?

[Steve Mann]

This may help, I just happen to be a google searching master :)

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

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of G.Marshall
Sent: Wednesday, April 06, 2005 11:06 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] ser <-> asterisk configs anyone?



I have searched high and low for these, but to no avail, nothing useful
back from google, nothing I could find on this mailing list, or
voip-user.org.

Does anyone have any good urls and or pointers which will assist in
configuring SIP Express Router and Asterisk talking to each other on the
same machine?

Many thanks,

Spencer

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[Asterisk-Users] RE: Asterisk and phone system

[Jason Kawakami]

-Original Message-

  Is there any way to get asterisk to wait 2 seconds before
it passes the rest of the phone number?  

-look at the w option to the dial command.

-Jason Kawakami

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Re: [Asterisk-Users] RE: Asterisk and phone system

[Eric Wieling aka ManxPower]

Jason Kawakami wrote:
-Original Message-

  Is there any way to get asterisk to wait 2 seconds before
it passes the rest of the phone number?  

-look at the w option to the dial command.
Dial(Zap/1/www1234) only works for ANALOG ports.  Look at the D() option 
in "show application dial".

--Eric
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Re: [Asterisk-Users] Stopping Retransmission Found: 102 Error with Polycom IP300

[Min Hwan Chang]

I'm also curious if this relates to qualify=yes.  
Incoming calls work fine with the IP300, I just can't dial out
currently.  When I look to check if any of my outgoing calls are in
the CDR, it is blank for outgoing calls. Only the incoming calls are
recorded.  I'll try messing with it  alittle bit more this morning.
Any other help would be appreciated.  Thank you all.

Regards,
Min 

On Apr 6, 2005 6:53 AM, Julian J. M. <[EMAIL PROTECTED]> wrote:
> I'm having this problem too, with a Swissvoice IP10... No nat between
> asterisk and the phone... I don't have any problems with the phone,
> outgoing and incoming calls work as expected...
> 
> Could it be related to qualify=yes?
> 
> Julian J. M.
> 
> On Apr 6, 2005 1:39 PM, Eric Wieling aka ManxPower <[EMAIL PROTECTED]> wrote:
> > Min Hwan Chang wrote:
> > > I'm having problems with a Polycom IP300 giving me a "Stopping
> > > Retransmission Found:102".  It gives this error about every 30
> > > seconds.
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[Asterisk-Users] Cant Hear Any Sound

[Ugur GUNCER]

I connect pri  to asterisk with e100p card when i call from pri i cant hear
any sound And when i call ip phone icant hear any sound. Does any one have
idea 


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Re: Re: [Asterisk-Users] Fritz Card ISDN in UK - Unable to

[Paul Redstone]

At the moment I think it is point to multi-point because:

1. It is an existing ISDN line used by an existing PABX.
2. We do not have DID - though plan to use this if our Asterisk trial is 
successful and we go with it.

We're also using the AVM chan_capi (current release 0.3.5) running on Debian 
Linux 2.4.18.

if we go live (which looks likely if I get get the demo going) we're thinking 
about the Junghans QuadBRI card, so interested in comments on this.

Paul

>>From: "Craig Guy" <[EMAIL PROTECTED]>
>>Subject: Re: [Asterisk-Users] Fritz Card ISDN in UK - Unable to
 >>   dial.0x3301/0x3302 errors
>>To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>>
>>Message-ID: <[EMAIL PROTECTED]>
>>Content-Type: text/plain;charset="iso-8859-1"
>>
>>Which fritz! driver are you using?  the AVM, or mISDN?  Also, are you
>>certain your line is in point to multipoint mode rather than point to point?
>>DID requires point to point, and at this time I'm not aware of anyone
>>getting it working with the Fritz! card.
>>
>>Craig
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Re: [Asterisk-Users] Web interface for realtime Mysql friends/peer

[Laurent Foulonneau]

Thanks
But  I was looking for a more complete solution like areski or astcc
Laurent
At 11:12 06/04/2005 +0200, Matteo Brancaleoni wrote:
phpmyadmin :)
Matteo.
Il giorno mer, 06-04-2005 alle 20:05 +1100, Laurent Foulonneau ha
scritto:
> Hello list,
>
> Does anyone know about a web/php interface to deal with users in 
Realtime's
> Mysql database (sipusers and sippeers tables) ?
>
> Thanks in advance
>
> Laurent
>
>

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[Asterisk-Users] Any success with BRI in the US?

[Patrick Conroy]

Hello all,

I have noticed a few people have mentioned recently that they are
looking to set up BRI in the US.  I am looking to do the same and I am
wondering if anyone has had any success yet?  And, if so what card are
you using?  I have heard that the Eicon Diva Server boards work, but
they are a little out of my price range at the moment.  Since
isdn4linux supports NI-1 (from what I have been reading) would any of
the passive BRI cards like the Fritz work?  Also, are there any boards
with U interface to avoid using an NT1 along with it?  Even the Eicon
Diva Server boards all seem to have the S/T interface.  Any
information would be helpful.

Thanks,
Patrick
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[Asterisk-Users] Script Perl Autodialer

[Maximiliano J. Goldsmid]

Hello everybody,

I am trying to use in a script perl, the originate action on a zap channel…..

$TelnetClient->print ("Action: Originate");
$TelnetClient->print ("Channel: Zap/$zap/$number");
$TelnetClient->print ("Context: out");
$TelnetClient->print ("Exten: s");
$TelnetClient->print ("Priority: 1");
$TelnetClient->print ("Variable: var1=$var1|var2=$var2");
$TelnetClient->print ("");
 

The problem is that when opening the zap channel, originate thinks
that the call has been answered and send the call to the beginning of
the context out. And what I really want is to make this but when the
destiny person answered and not when the zap channel opens.

 
So what can I do to solve it ou?
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[Asterisk-Users] Keypad disabled on AriaVoice SIP phone

[Ian Pattison]

Hi All,

I've got a couple of AriaVoice C-301P SIP phones connecting to * and have a bit 
of a problem. I can dial out with no problem but once * bridges the call to 
another channel (such as a Zap channel to the PSTN or an internal Analog phone) 
it appears that the keypad is then disabled which keeps me from navigating 
other peoples IVR trees. I've seen some ramblings on DTMF relay (usually when 
applied to Cisco products) but don't know if that's my issue at all. I've tried 
both RFC2833 and Inband Audio for DTMF handling with identical results. It 
literally appears that the keypad is being disabled after dialling. 

I've attempted to contact the vendor of the phones several times and have been 
unsuccessful in reaching anyone.

Anyone had a similar experience?

A snapshot of the phone's config can be seen at 
http://www.technologyassociates.ca/phone.jpg 

Thanks,

Ian

Ian Pattison, Senior Analyst
Technology Associates Inc.
Tel: 905-459-2100 ext. 204
Mobile: 416-568-6548
E-mail: [EMAIL PROTECTED] 

BEGIN:VCARD
VERSION:2.1
N:Pattison;Ian
FN:Ian Pattison
ORG:Technology Associates Inc.
ADR:;;9052 Creditview Rd.;Brampton;Ontario;L6V 1A1;Canada
TEL;WORK:416-657-2464
TEL;WORK:905-459-2100
TEL;CELL:416-568-6548
EMAIL:[EMAIL PROTECTED]
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Re: [Asterisk-Users] Script Perl Autodialer

[Adam Goryachev]

On Wed, 2005-04-06 at 14:28 -0300, Maximiliano J. Goldsmid wrote:
> Hello everybody,
> 
> I am trying to use in a script perl, the originate action on a zap 
> channelâ..
> 
> $TelnetClient->print ("Action: Originate");
> $TelnetClient->print ("Channel: Zap/$zap/$number");
> $TelnetClient->print ("Context: out");
> $TelnetClient->print ("Exten: s");
> $TelnetClient->print ("Priority: 1");
> $TelnetClient->print ("Variable: var1=$var1|var2=$var2");
> $TelnetClient->print ("");
>  
> 
> The problem is that when opening the zap channel, originate thinks
> that the call has been answered and send the call to the beginning of
> the context out. And what I really want is to make this but when the
> destiny person answered and not when the zap channel opens.
> 
>  
> So what can I do to solve it ou?

Use a channel that supports answer supervision, eg, ISDN/PRI/etc. Search
google for "asterisk answer supervision"

Maybe also attempt (if you are in the US) the callprogress=yes option.

Regards,
Adam
-- 
 -- 
Adam Goryachev
Website Managers
Ph:  +61 2 9345 4395[EMAIL PROTECTED]
Fax: +61 2 9345 4396www.websitemanagers.com.au

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[Asterisk-Users] I want to call another pc with TDM11B Card

[Yousri Farouk]

Hello
 
I have installed asterisk with card TDM11B 
successfully, now i want to call another pc has TDM11B Card and asterisk 
also.
 
please are there any one explain to me how can i do 
that.
 
thanks in advance
Regards 
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RE: [Asterisk-Users] RE: Asterisk and phone system

[Jeffrey Sharpe]

* is on a analog port of our legacy phone system which then converts it to
digital.

So do I use the W option?

 

Jeffrey Sharpe
CyberLynk Helpdesk and Support
414.858.9335 or 800.942.8022
[EMAIL PROTECTED]



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
aka ManxPower
Sent: Wednesday, April 06, 2005 11:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] RE: Asterisk and phone system

Jason Kawakami wrote:
> 
> -Original Message-
> 
>   Is there any way to get asterisk to wait 2 seconds before
> it passes the rest of the phone number?  
> 
> -look at the w option to the dial command.

Dial(Zap/1/www1234) only works for ANALOG ports.  Look at the D() option 
in "show application dial".

--Eric
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Re: [Asterisk-Users] Keypad disabled on AriaVoice SIP phone

[Jon Califf]

I had a simliar problem with my C-302P SIP phones until I added 
"dtmfmode=rfc2833" to my sip.conf

Ian Pattison wrote:
Hi All,
I've got a couple of AriaVoice C-301P SIP phones connecting to * and have a bit of a problem. I can dial out with no problem but once * bridges the call to another channel (such as a Zap channel to the PSTN or an internal Analog phone) it appears that the keypad is then disabled which keeps me from navigating other peoples IVR trees. I've seen some ramblings on DTMF relay (usually when applied to Cisco products) but don't know if that's my issue at all. I've tried both RFC2833 and Inband Audio for DTMF handling with identical results. It literally appears that the keypad is being disabled after dialling. 

I've attempted to contact the vendor of the phones several times and have been 
unsuccessful in reaching anyone.
Anyone had a similar experience?
A snapshot of the phone's config can be seen at http://www.technologyassociates.ca/phone.jpg 

Thanks,
Ian
Ian Pattison, Senior Analyst
Technology Associates Inc.
Tel: 905-459-2100 ext. 204
Mobile: 416-568-6548
E-mail: [EMAIL PROTECTED] 

 


BEGIN:VCARD
VERSION:2.1
N:Pattison;Ian
FN:Ian Pattison
ORG:Technology Associates Inc.
ADR:;;9052 Creditview Rd.;Brampton;Ontario;L6V 1A1;Canada
TEL;WORK:416-657-2464
TEL;WORK:905-459-2100
TEL;CELL:416-568-6548
EMAIL:[EMAIL PROTECTED]
END:VCARD
 


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Re: [Asterisk-Users] Script Perl Autodialer

[Brancaleoni Matteo]

Hi,

> The problem is that when opening the zap channel, originate thinks
> that the call has been answered and send the call to the beginning of
> the context out. And what I really want is to make this but when the
> destiny person answered and not when the zap channel opens.
> 
as already in the docs,
on analog zap interfaces you simply cannot do that,
since on analog there's no way (apart dsp) to guess
when the called party has answered

>  
> So what can I do to solve it ou?
go digital

(isdn bri/pri, voip, whatever)

Matteo
-- 

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RE: [Asterisk-Users] Keypad disabled on AriaVoice SIP phone

[Kanuri, Seshu (Company IT)]

Are these the same as YUXIN phones sold by eezeePhone.com? Do they have
PA1688 Chipset?

This does not look like a problem of the phones, but something to do
with Asterisk Dial plan. Are you using 'Answer' or 'Dial' command?

1)If you are usind Dial command, do not use T or t flags
2)DTMF mode Inband works only for Ulaw. If You use any other codecs, use
RFC2833 

Seshu


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ian
Pattison
Sent: Wednesday, April 06, 2005 1:34 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Keypad disabled on AriaVoice SIP phone

Hi All,

I've got a couple of AriaVoice C-301P SIP phones connecting to * and
have a bit of a problem. I can dial out with no problem but once *
bridges the call to another channel (such as a Zap channel to the PSTN
or an internal Analog phone) it appears that the keypad is then disabled
which keeps me from navigating other peoples IVR trees. I've seen some
ramblings on DTMF relay (usually when applied to Cisco products) but
don't know if that's my issue at all. I've tried both RFC2833 and Inband
Audio for DTMF handling with identical results. It literally appears
that the keypad is being disabled after dialling. 

I've attempted to contact the vendor of the phones several times and
have been unsuccessful in reaching anyone.

Anyone had a similar experience?

A snapshot of the phone's config can be seen at
http://www.technologyassociates.ca/phone.jpg 

Thanks,

Ian

Ian Pattison, Senior Analyst
Technology Associates Inc.
Tel: 905-459-2100 ext. 204
Mobile: 416-568-6548
E-mail: [EMAIL PROTECTED] 

 
NOTICE: If received in error, please destroy and notify sender.  Sender does 
not waive confidentiality or privilege, and use is prohibited. 
 
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Re: [Asterisk-Users] Keypad disabled on AriaVoice SIP phone

[Ian Pattison]

I've had that in there for a while now... no result for me.

Ian

>>> [EMAIL PROTECTED] 06/04/2005 13:46 >>>
I had a simliar problem with my C-302P SIP phones until I added 
"dtmfmode=rfc2833" to my sip.conf

Ian Pattison wrote:

>Hi All,
>
>I've got a couple of AriaVoice C-301P SIP phones connecting to * and have a 
>bit of a problem. I can dial out with no problem but once * bridges the call 
>to another channel (such as a Zap channel to the PSTN or an internal Analog 
>phone) it appears that the keypad is then disabled which keeps me from 
>navigating other peoples IVR trees. I've seen some ramblings on DTMF relay 
>(usually when applied to Cisco products) but don't know if that's my issue at 
>all. I've tried both RFC2833 and Inband Audio for DTMF handling with identical 
>results. It literally appears that the keypad is being disabled after 
>dialling. 
>
>I've attempted to contact the vendor of the phones several times and have been 
>unsuccessful in reaching anyone.
>
>Anyone had a similar experience?
>
>A snapshot of the phone's config can be seen at 
>http://www.technologyassociates.ca/phone.jpg 
>
>Thanks,
>
>Ian
>
>Ian Pattison, Senior Analyst
>Technology Associates Inc.
>Tel: 905-459-2100 ext. 204
>Mobile: 416-568-6548
>E-mail: [EMAIL PROTECTED] 
>
>  
>
>
>
>BEGIN:VCARD
>VERSION:2.1
>N:Pattison;Ian
>FN:Ian Pattison
>ORG:Technology Associates Inc.
>ADR:;;9052 Creditview Rd.;Brampton;Ontario;L6V 1A1;Canada
>TEL;WORK:416-657-2464
>TEL;WORK:905-459-2100
>TEL;CELL:416-568-6548
>EMAIL:[EMAIL PROTECTED] 
>END:VCARD
>
>  
>
>
>
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[Asterisk-Users] NOTICE: chan_sip:7654 handle_request: Registeration failed HELP !!

[Manjit Riat]

Hi,

   I have a
setup asterisk with 4 extensions with asterisk & 3 extensions on public IP
and one extension behind a NAT. All works great. Today I tried installing X-lite for my friends in Dubai (ISP uses a proxy server/NAT) and I get
this error

 

Apr  6 10:51:41
NOTICE[1213]: chan_sip.c:7654 handle_request:
Registration from 'Someone ' failed for
'217.xxx.xxx.xxx'

 

I have this in sip.conf

 

[someone]

type=friend

host=dynamic

nat=yes

dtmfmode=inband

username=someone

secret=someone

context=sip

callerid="Someone"
<5>

mailbox=5

disallow=all

allow=ulaw

 

 

Is that a problem on my end or config
problems on the X-lite?

 

I know someone had a extension from
Dubai working. So
could someone help me.

 

Thank You

 






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Re: [Asterisk-Users] Web interface for realtime Mysql friends/peer

[G.Marshall]

> Thanks
>
> But  I was looking for a more complete solution like areski or astcc
I found nothing so I wrote my own, but they are for postgres.  They are
not complete by no means.  If you are interested, I will let you have a
look at what I have done, and if you provide constructive critisism, I
will be happy to release the php under the same licence as Asterisk.

>
> Laurent
> At 11:12 06/04/2005 +0200, Matteo Brancaleoni wrote:
>
>>phpmyadmin :)
>>
>>Matteo.
>>
>>Il giorno mer, 06-04-2005 alle 20:05 +1100, Laurent Foulonneau ha
>>scritto:
>> > Hello list,
>> >
>> > Does anyone know about a web/php interface to deal with users in
>> Realtime's
>> > Mysql database (sipusers and sippeers tables) ?
>> >
>> > Thanks in advance
>> >

>> > Laurent
>> >
>> >
>>
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RE: [Asterisk-Users] ser <-> asterisk configs anyone?

[G.Marshall]

> This may help, I just happen to be a google searching master :)
I bow to the master ;) seriously though, thank you.

>
> http://www.voip-info.org/wiki-Asterisk+at+large
I thought I had seen something somewhere.  Plus I miss quoted
voip-user.org for voip-info.org

Thank you

>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of G.Marshall
> Sent: Wednesday, April 06, 2005 11:06 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] ser <-> asterisk configs anyone?
>
>
>
> I have searched high and low for these, but to no avail, nothing useful
> back from google, nothing I could find on this mailing list, or
> voip-user.org.
>
> Does anyone have any good urls and or pointers which will assist in
> configuring SIP Express Router and Asterisk talking to each other on the
> same machine?
>
> Many thanks,
>
> Spencer
>
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RE: [Asterisk-Users] Keypad disabled on AriaVoice SIP phone

[Ian Pattison]

I'm not sure about the chipset but they do appear to be similar... with the 
exception of IAX2 support which mine do not have, SIP, H.323 and MGCP only.

A bit more detail that I've uncovered... if I call into asterisk from an 
outside line and connect to my SIP phone I get DTMF tones passed in both 
directions. If I call from my SIP phone to an internal Zap extension DTMF is 
passed properly. But if I call from my SIP phone to an external phone (via a 
ZAP FXO channel) the keypad shuts down and nothing is passed. Here's the 
important part of my extensions.conf:

exten => _9NXXNXX,1,Dial(Zap/2/${EXTEN:1})

Thanks,

Ian

>>> [EMAIL PROTECTED] 06/04/2005 13:57 >>>
Are these the same as YUXIN phones sold by eezeePhone.com? Do they have
PA1688 Chipset?

This does not look like a problem of the phones, but something to do
with Asterisk Dial plan. Are you using 'Answer' or 'Dial' command?

1)If you are usind Dial command, do not use T or t flags
2)DTMF mode Inband works only for Ulaw. If You use any other codecs, use
RFC2833 

Seshu


-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Ian
Pattison
Sent: Wednesday, April 06, 2005 1:34 PM
To: asterisk-users@lists.digium.com 
Subject: [Asterisk-Users] Keypad disabled on AriaVoice SIP phone

Hi All,

I've got a couple of AriaVoice C-301P SIP phones connecting to * and
have a bit of a problem. I can dial out with no problem but once *
bridges the call to another channel (such as a Zap channel to the PSTN
or an internal Analog phone) it appears that the keypad is then disabled
which keeps me from navigating other peoples IVR trees. I've seen some
ramblings on DTMF relay (usually when applied to Cisco products) but
don't know if that's my issue at all. I've tried both RFC2833 and Inband
Audio for DTMF handling with identical results. It literally appears
that the keypad is being disabled after dialling. 

I've attempted to contact the vendor of the phones several times and
have been unsuccessful in reaching anyone.

Anyone had a similar experience?

A snapshot of the phone's config can be seen at
http://www.technologyassociates.ca/phone.jpg 

Thanks,

Ian

Ian Pattison, Senior Analyst
Technology Associates Inc.
Tel: 905-459-2100 ext. 204
Mobile: 416-568-6548
E-mail: [EMAIL PROTECTED] 

 
NOTICE: If received in error, please destroy and notify sender.  Sender does 
not waive confidentiality or privilege, and use is prohibited. 
 
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RE: [Asterisk-Users] Web interface for realtime Mysql friends/peer

[Kanuri, Seshu (Company IT)]

Marshall,

I am interested in seeing what you wrote to manage MySQL database
objects. 

By the way, latest version of OpenOffice comes with a MySQL
Administrator GUI to manage tables and data. This is something to look
at too.

Seshu Kanuri


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of G.Marshall
Sent: Wednesday, April 06, 2005 2:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Web interface for realtime Mysql
friends/peer

> Thanks
>
> But  I was looking for a more complete solution like areski or astcc
I found nothing so I wrote my own, but they are for postgres.  They are
not complete by no means.  If you are interested, I will let you have a
look at what I have done, and if you provide constructive critisism, I
will be happy to release the php under the same licence as Asterisk.

>
> Laurent
> At 11:12 06/04/2005 +0200, Matteo Brancaleoni wrote:
>
>>phpmyadmin :)
>>
>>Matteo.
>>
>>Il giorno mer, 06-04-2005 alle 20:05 +1100, Laurent Foulonneau ha
>>scritto:
>> > Hello list,
>> >
>> > Does anyone know about a web/php interface to deal with users in
>> Realtime's
>> > Mysql database (sipusers and sippeers tables) ?
>> >
>> > Thanks in advance
>> >

>> > Laurent
>> >
>> >
>>
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>>
>>
>>--
>>No virus found in this incoming message.
>>Checked by AVG Anti-Virus.
>>Version: 7.0.308 / Virus Database: 266.9.3 - Release Date: 05/04/2005
>
>
> --
> No virus found in this outgoing message.
> Checked by AVG Anti-Virus.
> Version: 7.0.308 / Virus Database: 266.9.3 - Release Date: 05/04/2005
>
>
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Re: [Asterisk-Users] Automatic Start

[Cameron Beattie]

I'm no Linux guru but my understanding is that the script you are referring 
to (/etc/rc3.d/S40asterisk) just points to the real script in 
/etc/rc.d/init.d or whereever (depends on which linux you're using). The rcn 
(e.g. rc3) refers to the run level i.e. 3 in your case. So when you boot 
linux in run level 3 it looks in rc3 and runs the scripts there in numerical 
order (asterisk script is number 40 in priority)- the S indicates start.

Anyway, you should modify the script in init.d.
Regards
Cameron
- Original Message - 
From: "J Hobbs" <[EMAIL PROTECTED]>
To: 
Sent: Wednesday, April 06, 2005 3:36 PM
Subject: Re: [Asterisk-Users] Automatic Start


Thank you for your reply, David.
I found the file in /etc/rc3.d/S40asterisk. Where would I put the line to 
load the Wildcard PCI interface card (modprobe wcfxs)?

Would I put it before the line:
RETVAL = 0
start() {
 # start daemons
 etc
}

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

To: Asterisk Users Mailing List - Non-Commercial Discussion 

CC: 
asterisk-users@lists.digium.com,[EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Automatic Start
Date: Wed, 6 Apr 2005 10:21:55 +0800

Hi,
If its started from an automated script, it should be in
/etc/rc.d/init.d/asterisk
If you want a quick and easy fix to start asterisk after zaptel do this
(Assuming that Zaptel is installed properly)
Modify /etc/rc.local
Add the following lines.
service asterisk stop
service zaptel restart
service asterisk start
It will first start asterisk, then restart zaptel (to ensure that even if
its started, it will work) then start asterisk
Best Regards,
==
David Choo
Systems Engineer
Business & Technology Division
"Engineered for Changing Businesses"
Espore Corp Pte Ltd
68 Kallang Pudding Rd
#04-03 SYH Logistics Bldg
Singapore 349327
Tel: 65-68487806
Fax : 65-6842 2724
E-mail :[EMAIL PROTECTED]
=
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Internet communications cannot be guaranteed to be secure or error-free as
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viruses. The sender therefore does not accept liability for any errors or
omissions in the context of this message nor can the sender guarantee that
this message is virus free.

 "J Hobbs"
 <[EMAIL PROTECTED]
 il.com> 
To
 Sent by:  asterisk-users@lists.digium.com
 asterisk-users-bo 
cc
 [EMAIL PROTECTED]
 m.com 
Subject
   [Asterisk-Users] Automatic Start

 06/04/2005 08:17
 AM
 Please respond to
  Asterisk Users
  Mailing List -
  Non-Commercial
Discussion
 <[EMAIL PROTECTED]
 ists.digium.com>


Hi All,
I have just installed Asterisk using Signate install disks which installed
CentOS (Redhat) and then loaded Asterisk.
I am completely new to Linux so this is probably a dumb question, but I
can't figure it out or find any documents that refer to it. So here goes.
When I start CentOS it automatically loads Asterisk and Gnome, but where 
is

the startup configuration file for Asterisk. I need to edit this file so
that the Wildcard is also loaded.
Put me out of my misery please.
_
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[Asterisk-Users] SIP - SIP Problems

[Ian Pattison]

Hi Everybody...

Continuing the litany of problems I'm experiencing with my new system I'm 
getting issues connecting between SIP phones.

A bit of background... I have an asterisk server running in a central location 
where I have two incoming analog lines connected to FXO ports, two analog 
phones connecting to FXS ports and a single SIP phone. In addition I have a 
remote site connected via a CIPE VPN (ok..ok I know it's not a real VPN...) 
with a single SIP phone. 

Calls initiated from the remote SIP phone to the central SIP phone often have 
trouble... the user of the central phone cannot hear anything from the remote 
phone although everything is heard at the remote phone. If the remote phone 
calls either outside or to one of the Zap phones there is no trouble. If the 
local SIP phone calls the remote SIP phone there is no trouble. Both phones are 
from the same vendor, have the same firmware and the same configuration with 
the exception of phone number, PIN, IP address etc.

What am I doing wrong here?

Ian

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[Asterisk-Users] Connecting asterisk to existing PBX - newbie question

[Dan Reagan]

I have a question regarding interoperability between Asterisk and an 
Inter-Tel (I believe it's an Axxess) PBX at a client's location.

My clients currently have an Inter-Tel system that they're fairly happy 
with but they're being bludgeoned by Inter-Tel for their proprietary IP 
based phones. The phones themselves are on the order of $1,000 each 
(they're in a lease and only see the monthly charge when they add a new 
phone. That $1000 is a rough approximation on my part from the last 
lease addition which was $24/month with 42 months remaining).

I believe that many of the people that they have who are currently using 
the proprietary phones would be quite happy with a basic SIP phone (they 
don't need ACD login or other advanced features) however Inter-Tel's SIP 
gateway product is quite expensive. The sales person never actually gave 
me a real quote but said things like "pushing $10,000" and "very 
expensive" and "You don't really want it". My feeling was that they want 
to keep this client with their proprietary phones (and at $1000 each I 
can see why).

I think that I could craft a gateway of some sort out of an Asterisk box 
doing nothing but taking a handoff from the Inter-Tel PBX and passing it 
out to SIP phones attached to the Asterisk machine but I'm not 
absolutely certain how to go about it. Off the top of my head I see two 
rough directions to go but I'm somewhat out of my depth and would 
appreciate any insight, particularly with regard to the details .

First, I could take an existing 'single line card' in the Inter-Tel 
system (8 FXS ports already installed in the existing PBX) and run that 
into 8 FXO ports on the Asterisk machine. I'm assuming that from there 
it would be a fairly straightforward to set up a 1 to 1 mapping of 
incoming FXO to connected SIP phones.

Second, I could take an existing T1 card (I'd have to buy an upgrade to 
get PRI functionality if required) and connect it to the Asterisk box. 
At first glance this seems to be a more flexible and better solution but 
I believe that on the Inter-Tel side I'd have some issues with mapping 
extensions to the outgoing PRI ports and/or extensions on the asterisk 
side and a fair amount of jiggling to get things working right.

Alternatively, would it be possible to just stick something like the 
AudioCodes MP108-FXO in the middle of the 8 FXS ports on the Inter-Tel 
PBX and some SIP phones? Initially I thought that this would work but 
further reading has led me to think that this wouldn't work and that's 
what led me back to Asterisk.

That's my very rough thinking. Any help would be greatly appreciated. 
Also, I'm sorry if this has been covered before on the list or 
elsewhere. I spent some time digging through the archives and couldn't 
find anything helpful but I'm sure that I wasn't perfectly thorough.

Dan Reagan
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Re: [Asterisk-Users] asterisk is giving error- unable to write audiodata codec_speex.so

[Cameron Beattie]

Have you looked in the log file? Might be more info' there
Regards
Cameron
- Original Message - 
From: <[EMAIL PROTECTED]>
To: 
Sent: Wednesday, April 06, 2005 10:30 PM
Subject: [Asterisk-Users] asterisk is giving error- unable to write 
audiodata codec_speex.so


hi friends !
my asterisk is giving one error while running.
it says that unable to write audio data, module codec_speex.so is not 
loaded.
have anybody face this kind of problem than plz tell me the solution.
thanks
Deepak Dhiman
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[Asterisk-Users] SRV Bounty

[Matt Schulte]

Is there an SRV bounty out there yet? $500 to the first person who
implements it (correctly :-) )..

Email for details.
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Re: [Asterisk-Users] Asterisk and phone system

[Cameron Beattie]

You could try having a look at the CLI (asterisk 
-rv) to ensure that Asterisk is doing what you think it is.
 
Regards
 
Cameron

  - Original Message - 
  From: 
  Jeffrey 
  Sharpe 
  To: 'Asterisk Users Mailing List - 
  Non-Commercial Discussion' 
  Sent: Thursday, April 07, 2005 3:00 
  AM
  Subject: [Asterisk-Users] Asterisk and 
  phone system
  
  
  I have my * test box connected to 
  our phone system here on an analog port with a X100P 
  card.
   
  The phone system requires a 9 to 
  pickup an outside line.  Problem is, I think asterisk is passing the 
  phone number to quickly after the 9 and not getting the line.  Is there 
  any way to get asterisk to wait 2 seconds before it passes the rest of the 
  phone number?  We have had this problem with a fax machine before, and 
  had to set it up to pause after the 9 for 2 
  seconds.
   
  The lines into the building are 
  centrex if that makes a difference. 
   
   
  Jeffrey 
  SharpeCyberLynk Helpdesk and Support414.858.9335 or 
  800.942.8022[EMAIL PROTECTED]
   
  
  

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[Asterisk-Users] Problems using Asterisk 1.0.3 with Vocal 1.4

[Daniel Mizrachi (Globalsip)]

Hi...
 
Since I upgraded the asterisk version from 0.93 to 
1.0.3 I have had a lot of problems to complete calls thru Vocal Sip Proxy 
1.4. 
 
When I just register Asterisk to receive calls from 
vocal it works fine. Ex.
 
register => 
2127701002:[EMAIL PROTECTED]/2098
 
And
 
When I just regiter asterisk to send call thru 
Vocal it also works fine. Ex.
 
[sip-proxy-out]type=peer   
; we only want to call out, not be 
calledsecret=paswordusername=2127701005    
; Authentication user for outbound 
proxiesfromuser=2127701005 
; Many SIP providers require 
this!fromdomain=domain.comhost=mysipproxy.domain.comnat=yesinsecure=yes
The problem is when I try to use either incoming or 
outgoing calls at the same time. For Any reason the incoming calls don´t work 
and the sip messages that I got are:
 
Sip read:ACK sip:[EMAIL PROTECTED] 
SIP/2.0Via: SIP/2.0/UDP 
210.80.130.84:5060;rport=5060;received=210.80.132.86;branch=z9hG4bK61f8896885a9fa343f3bc812cb933e99.4To: 
;tag=as0e7e00ebFrom: 
"2127100139"Call-ID: [EMAIL PROTECTED]CSeq: 
101 ACKMax-Forwards: 70Content-Length: 0
 
8 headers, 0 linesApr  5 07:55:18 NOTICE[10601]: 
chan_sip.c:4007 sip_reregister:    -- Re-registration for  
[EMAIL PROTECTED]12 
headers, 0 linesReliably Transmitting:REGISTER sip:mysipproxy.domain.com 
SIP/2.0Via: SIP/2.0/UDP 210.80.132.86:5060;branch=z9hG4bK4b215889From: 
;tag=as6c056f3bTo: 
Call-ID: [EMAIL PROTECTED]CSeq: 
104 REGISTERUser-Agent: Asterisk PBXAuthorization: Digest 
username="2127701002", realm="210.80.130.84", algorithm=MD5, 
uri="210.80.130.84", nonce="1112588169", 
response="73f1564125d54c9982ec70a90286", opaque=""Expires: 
120Contact: Event: 
registrationContent-Length: 0
 
 (no NAT) to 210.80.130.84:5060lab-ing*CLI>
 
Sip read:SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 
210.80.132.86:5060;branch=z9hG4bK4b215889To: 
;tag=df08f10cFrom: 
;tag=as6c056f3bCall-ID: [EMAIL PROTECTED]CSeq: 
104 REGISTERProxy-Authenticate: Digest 
algorithm=MD5,domain=210.80.130.84,nonce=1113026437,realm=210.80.130.84Content-Length: 
0
 
8 headers, 0 linesResponding to challenge, registration to 
domain/host name mysipproxy.domain.com12 headers, 0 linesReliably 
Transmitting:REGISTER sip:mysipproxy.domain.com SIP/2.0Via: SIP/2.0/UDP 
210.80.132.86:5060;branch=z9hG4bK6cac2088From: 
;tag=as6c056f3bTo: 
;tag=df08f10cCall-ID: [EMAIL PROTECTED]CSeq: 
105 REGISTERUser-Agent: Asterisk PBXProxy-Authorization: Digest 
username="2127701002", realm="210.80.130.84", algorithm=MD5, 
uri="210.80.130.84", nonce="1113026437", 
response="a173d7c8a6874f697c8dafb2d48eed13", opaque=""Expires: 
120Contact: Event: 
registrationContent-Length: 0
 
 (no NAT) to 210.80.130.84:5060Retransmitting #1 (no 
NAT):REGISTER sip:mysipproxy.domain.com SIP/2.0Via: SIP/2.0/UDP 
210.80.132.86:5060;branch=z9hG4bK6cac2088From: 
;tag=as6c056f3bTo: 
;tag=df08f10cCall-ID: [EMAIL PROTECTED]CSeq: 
105 REGISTERUser-Agent: Asterisk PBXProxy-Authorization: Digest 
username="2127701002", realm="210.80.130.84", algorithm=MD5, 
uri="210.80.130.84", nonce="1113026437", 
response="a173d7c8a6874f697c8dafb2d48eed13", opaque=""Expires: 
120Contact: Event: 
registrationContent-Length: 0
 
 to 210.80.130.84:5060lab-ing*CLI>
 
Sip read:SIP/2.0 100 TryingVia: SIP/2.0/UDP 
210.80.132.86:5060;branch=z9hG4bK6cac2088To: 
;tag=df08f10cFrom: 
;tag=as6c056f3bCall-ID: [EMAIL PROTECTED]CSeq: 
105 REGISTERContent-Length: 0
 
7 headers, 0 lineslab-ing*CLI>
 
Sip read:SIP/2.0 200 OKVia: SIP/2.0/UDP 
210.80.132.86:5060;branch=z9hG4bK6cac2088To: 
;tag=df08f10cFrom: 
;tag=as6c056f3bCall-ID: [EMAIL PROTECTED]CSeq: 
105 REGISTERExpires: 120Contact: 
Content-Length: 0
 
9 headers, 0 linesApr  5 07:55:19 NOTICE[10601]: 
chan_sip.c:6797 handle_response: Outbound Registration: Expiry for 
mysipproxy.domain.com is 120 sec (Scheduling reregistration in 105000 
ms)
 
 Thanks in advance for any support to solve 
it.
 
 
Best Regards,
 
 
Danil Mizrachi
Globalsip
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[Asterisk-Users] Ingate Firewall and Asterisk Integration

[c waddy]

Hi,

I just got an Ingate Firewall, I cannot get asterisk to register to
it. Using UDP port forwarding to the Asterisk box, its like the
firewall is re-writing the password part.

I keep getting an error like the password is wrong, when trying to
route through the firewall.

 In the console i get.

WARNING[19393]: chan_sip.c:6834 handle_response: Forbidden - wrong
password on authentication for REGISTER for 'one' to '195.***.22.12'

Yet i know the password is correct.

In sip.conf  -> register => test:[EMAIL PROTECTED]:5060

Anyone had asterisk + Ingate working?

Is there a certain authentication method or encryption i need to use?

Below is the log output showing asterisk trying to authenticate with the Ingate.

UDP   195.224.22.12 5060 eth1 10.6.0.9 5060 Accepted SIP signaling
2005-04-04 15:55:27 >>> Info: sipfw:  send sl to 10.6.0.9:

 SIP/2.0 403 Forbidden
 Via: SIP/2.0/UDP 10.6.0.9:5060;branch=z9hG4bK0f09ac29
 From: ;tag=as576cb37b
 Call-ID: [EMAIL PROTECTED]
 CSeq: 230 REGISTER
 Server: Ingate-Firewall/4.1.3
 To: ;tag=10675194461474
 Content-Length: 0

2005-04-04 15:55:27 >>> Info: sipfw: send sl to 10.6.0.9: SIP/2.0 403 Forbidden
2005-04-04 15:55:27 >>> Notice: sipfw: REGISTER request denied.
2005-04-04 15:55:27 >>> Info: sipfw:  recv from 10.6.0.9 via socket 12:

 REGISTER sip:195.***.22.12 SIP/2.0
 Via: SIP/2.0/UDP 10.6.0.9:5060;branch=z9hG4bK0f09ac29
 From: ;tag=as576cb37b
 To: 
 Call-ID: [EMAIL PROTECTED]
 CSeq: 230 REGISTER
 User-Agent: Asterisk PBX
 Expires: 120
 Contact: 
 Event: registration
 Content-Length: 0

2005-04-04 15:55:27 >>> Info: sipfw: recv from 10.6.0.9: REGISTER
sip:195.224.22.12 SIP/2.0
2005-04-04 15:55:27.703 UDP eth1 10.6.0.9 5060   195.***.22.12 5060
Accepted SIP signaling
2005-04-04 15:55:07.688 UDP   195.***.22.12 5060 eth1 10.6.0.9 5060
Accepted SIP signaling
2005-04-04 15:55:07 >>> Info: sipfw:  send sl to 10.6.0.9:

 SIP/2.0 403 Forbidden
 Via: SIP/2.0/UDP 10.6.0.9:5060;branch=z9hG4bK3d04eca3
 From: ;tag=as0e819f3b
 Call-ID: [EMAIL PROTECTED]
 CSeq: 229 REGISTER
 Server: Ingate-Firewall/4.1.3
 To: ;tag=10675194461474
 Content-Length: 0
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Re: [Asterisk-Users] wcte11xp works only after cold reboot

[Ron Wellsted]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Stuart Hirst wrote:
> Alessio,
> 
> I have also seen this problem on two different asterisk servers using
> TDM400p cards.
> 
> I have not been able to resolve it. If you do an lspci you can see that
> the system can see the devices but the zaptel drivers don't see them.
> 
> I have other systems that work fine and so this has to be down to the
> combination of motherboard and Digium PCI devices. It would seem that
> this is a problem with the the Digium hardware because it works after a
> cold boot.
> 
> Does anyone else have a view ?
> 
> Stuart
> 
> Stuart
> 
> On Wed, 2005-04-06 at 12:26 +0200, Alessio Focardi wrote:
> 
>>Hi,
>>
>>my brand new wcte11xp works like a charme of first boot, then if I
>>
>>shutdown -r now
>>
>>the server is not detected at reboot ("no such device" after modprobe).
>>
>>Turning off the pc and cold restarting fixes the problem.
>>
>>Has someone experienced such behaviour before ?
>>
>>Tnx for any help!

This is a known issue and crops up on the list from time to time.  For
some reason, the card will change is PCI bus ID.  It is set back to the
correct ID by a cold reboot.

To work around this, edit the wcte1xxp.c in the zaptel directory so that
the pci_device_id structure becomes:

static struct pci_device_id t1xxp_pci_tbl[] = {
{ 0xe159, 0x0001, 0x71fe, PCI_ANY_ID, 0, 0, (unsigned long)
"Digium Wildcard TE110P T1/E1 Board" },
{ 0xe159, 0x0001, 0x79fe, PCI_ANY_ID, 0, 0, (unsigned long)
"Digium Wildcard TE110P T1/E1 Board" },
{ 0xe159, 0x0001, 0x795e, PCI_ANY_ID, 0, 0, (unsigned long)
"Digium Wildcard TE110P T1/E1 Board" },
{ 0xe159, 0x0001, 0x79de, PCI_ANY_ID, 0, 0, (unsigned long)
"Digium Wildcard TE110P T1/E1 Board" },
{ 0xe159, 0x0001, 0x797e, PCI_ANY_ID, 0, 0, (unsigned long)
"Digium Wildcard TE110P T1/E1 Board" },
{ 0 }
};

This should allow reboots but YMMV (well, it works on my server!).

HTH

- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961  Gossiptel:9309811
N 52.567623, W 2.137621
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[Asterisk-Users] Asterisk .call files

[Gilbert Abboud]

hi

I created a .call file as mentioned in the WiKi but when i place it in 
/var/spool/asterisk/outgoing, the Asterisk console shows "unknown keyword" for 
all the keywords used in the .call file (i.e channel, context, extension,...). 
Any ideas why?

Regards,

Gilbert Abboud
M.Eng. Computer Engineering
Programmer Analyst
Excendia, Montreal
ESN: 514-765-8490

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RE: [Asterisk-Users] Keypad disabled on AriaVoice SIP phone

[Kanuri, Seshu (Company IT)]

Let us try the opposite of what I have suggested and see what it does.
Change the Dial command as under and see how that goes.

exten => _9NXXNXX,1,Dial(Zap/2/${EXTEN:1},30, Tt)

Seshu

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ian
Pattison
Sent: Wednesday, April 06, 2005 2:38 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Keypad disabled on AriaVoice SIP phone

I'm not sure about the chipset but they do appear to be similar... with
the exception of IAX2 support which mine do not have, SIP, H.323 and
MGCP only.

A bit more detail that I've uncovered... if I call into asterisk from an
outside line and connect to my SIP phone I get DTMF tones passed in both
directions. If I call from my SIP phone to an internal Zap extension
DTMF is passed properly. But if I call from my SIP phone to an external
phone (via a ZAP FXO channel) the keypad shuts down and nothing is
passed. Here's the important part of my extensions.conf:

exten => _9NXXNXX,1,Dial(Zap/2/${EXTEN:1})

Thanks,

Ian

>>> [EMAIL PROTECTED] 06/04/2005 13:57 >>>
Are these the same as YUXIN phones sold by eezeePhone.com? Do they have
PA1688 Chipset?

This does not look like a problem of the phones, but something to do
with Asterisk Dial plan. Are you using 'Answer' or 'Dial' command?

1)If you are usind Dial command, do not use T or t flags 2)DTMF mode
Inband works only for Ulaw. If You use any other codecs, use
RFC2833 

Seshu


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ian
Pattison
Sent: Wednesday, April 06, 2005 1:34 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Keypad disabled on AriaVoice SIP phone

Hi All,

I've got a couple of AriaVoice C-301P SIP phones connecting to * and
have a bit of a problem. I can dial out with no problem but once *
bridges the call to another channel (such as a Zap channel to the PSTN
or an internal Analog phone) it appears that the keypad is then disabled
which keeps me from navigating other peoples IVR trees. I've seen some
ramblings on DTMF relay (usually when applied to Cisco products) but
don't know if that's my issue at all. I've tried both RFC2833 and Inband
Audio for DTMF handling with identical results. It literally appears
that the keypad is being disabled after dialling. 

I've attempted to contact the vendor of the phones several times and
have been unsuccessful in reaching anyone.

Anyone had a similar experience?

A snapshot of the phone's config can be seen at
http://www.technologyassociates.ca/phone.jpg 

Thanks,

Ian

Ian Pattison, Senior Analyst
Technology Associates Inc.
Tel: 905-459-2100 ext. 204
Mobile: 416-568-6548
E-mail: [EMAIL PROTECTED]

 
NOTICE: If received in error, please destroy and notify sender.  Sender
does not waive confidentiality or privilege, and use is prohibited. 
 
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RE: [Asterisk-Users] Asterisk .call files

[Alexander Lopez]

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gilbert
Abboud
Sent: Wednesday, April 06, 2005 2:47 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk .call files

hi

I created a .call file as mentioned in the WiKi but when i place it in
/var/spool/asterisk/outgoing, the Asterisk console shows "unknown
keyword" for all the keywords used in the .call file (i.e channel,
context, extension,...). 
Any ideas why?

Regards,

Gilbert Abboud
M.Eng. Computer Engineering
Programmer Analyst
Excendia, Montreal
ESN: 514-765-8490

___


post your .call file so that we can see..
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RE: [Asterisk-Users] Asterisk .call files

[Kanuri, Seshu (Company IT)]

Remove all the spaces in front of the lines

Seshu


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gilbert
Abboud
Sent: Wednesday, April 06, 2005 3:47 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk .call files

hi

I created a .call file as mentioned in the WiKi but when i place it in
/var/spool/asterisk/outgoing, the Asterisk console shows "unknown
keyword" for all the keywords used in the .call file (i.e channel,
context, extension,...). 
Any ideas why?

Regards,

Gilbert Abboud
M.Eng. Computer Engineering
Programmer Analyst
Excendia, Montreal
ESN: 514-765-8490

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[Asterisk-Users] Asterisk, ACD, Queues and Call Transfer Issue

[Dov Bigio]

Hello,
 
I have implemented a test ACD with Asterisk 1.0.7, in which I have 2 agents and one user making calls and using AgentCallbackLogin. Besides that, I have other users on the PBX, but not necessarily members of any queue.
 
Agents and PBX users are using X-Pro as a soft-phone.
 
I am having problems in the case where one agent answers the call and for any reason needs to transfer this call to the other agent, or to somebody else in the PBX system.
 
When the agent clicks the transfer button on the soft-phone, the call is hang up and we loose the client. If the agent dials '#' then the transfer works fine. The problem is that in this case the person to whom the call is being transferred must have a numerical extension (which we didn't want to use internally).
 
is there a solution for this?
 
Thank you
Dov
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Re: [Asterisk-Users] Asterisk .call files

[Andrew Kohlsmith]

On April 6, 2005 03:47 pm, Gilbert Abboud wrote:
> I created a .call file as mentioned in the WiKi but when i place it in
> /var/spool/asterisk/outgoing, the Asterisk console shows "unknown keyword"
> for all the keywords used in the .call file (i.e channel, context,
> extension,...). Any ideas why?

http://www.catb.org/~esr/faqs/smart-questions.html

Give us some details (hell the .callfile would be handy perhaps) and come 
back.  We can't help you if you won't give us the information we need to 
assist.

-A.
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Re: [Asterisk-Users] Asterisk .call files

[João Amaro]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi.

If you show what's in the .call file, it would help.

Rgs

Joao


Gilbert Abboud wrote:

| hi
|
| I created a .call file as mentioned in the WiKi but when i place it
| in /var/spool/asterisk/outgoing, the Asterisk console shows
| "unknown keyword" for all the keywords used in the .call file (i.e
| channel, context, extension,...). Any ideas why?
|
| Regards,
|
| Gilbert Abboud M.Eng. Computer Engineering Programmer Analyst
| Excendia, Montreal ESN: 514-765-8490
|
| ___ Asterisk-Users
| mailing list Asterisk-Users@lists.digium.com
| http://lists.digium.com/mailman/listinfo/asterisk-users To
| UNSUBSCRIBE or update options visit:
| http://lists.digium.com/mailman/listinfo/asterisk-users
|
|
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Version: GnuPG v1.2.4 (GNU/Linux)

iD8DBQFCVDy8JUm/Bor63CERAtL/AJ9l03ibSF8bdMtDrD1sLo6DF3GZwQCeNa1Z
dM68KlX8Hnx2xqfn2eIy7ko=
=jQMN
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[Asterisk-Users] RE:RE: RE: Asterisk and phone system

[Jason Kawakami]

-Original Message-

   * is on a analog port of our legacy phone system which then converts it
to digital.

So do I use the W option?

-per your original post, on an x100 you would use the
Dial(zap/1/www${EXTEN})

-Jason Kawakami

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RE: [Asterisk-Users] Keypad disabled on AriaVoice SIP phone

[Ian Pattison]

No change whatsoever. I still think it's something with the phone... none of 
the keys have any effect after dialling... they do not echo to the display 
even. It's literally as though the keypad has been disabled.

Ian

>>> [EMAIL PROTECTED] 06/04/2005 15:41 >>>
Let us try the opposite of what I have suggested and see what it does.
Change the Dial command as under and see how that goes.

exten => _9NXXNXX,1,Dial(Zap/2/${EXTEN:1},30, Tt)

Seshu

-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Ian
Pattison
Sent: Wednesday, April 06, 2005 2:38 PM
To: asterisk-users@lists.digium.com 
Subject: RE: [Asterisk-Users] Keypad disabled on AriaVoice SIP phone

I'm not sure about the chipset but they do appear to be similar... with
the exception of IAX2 support which mine do not have, SIP, H.323 and
MGCP only.

A bit more detail that I've uncovered... if I call into asterisk from an
outside line and connect to my SIP phone I get DTMF tones passed in both
directions. If I call from my SIP phone to an internal Zap extension
DTMF is passed properly. But if I call from my SIP phone to an external
phone (via a ZAP FXO channel) the keypad shuts down and nothing is
passed. Here's the important part of my extensions.conf:

exten => _9NXXNXX,1,Dial(Zap/2/${EXTEN:1})

Thanks,

Ian

>>> [EMAIL PROTECTED] 06/04/2005 13:57 >>>
Are these the same as YUXIN phones sold by eezeePhone.com? Do they have
PA1688 Chipset?

This does not look like a problem of the phones, but something to do
with Asterisk Dial plan. Are you using 'Answer' or 'Dial' command?

1)If you are usind Dial command, do not use T or t flags 2)DTMF mode
Inband works only for Ulaw. If You use any other codecs, use
RFC2833 

Seshu


-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Ian
Pattison
Sent: Wednesday, April 06, 2005 1:34 PM
To: asterisk-users@lists.digium.com 
Subject: [Asterisk-Users] Keypad disabled on AriaVoice SIP phone

Hi All,

I've got a couple of AriaVoice C-301P SIP phones connecting to * and
have a bit of a problem. I can dial out with no problem but once *
bridges the call to another channel (such as a Zap channel to the PSTN
or an internal Analog phone) it appears that the keypad is then disabled
which keeps me from navigating other peoples IVR trees. I've seen some
ramblings on DTMF relay (usually when applied to Cisco products) but
don't know if that's my issue at all. I've tried both RFC2833 and Inband
Audio for DTMF handling with identical results. It literally appears
that the keypad is being disabled after dialling. 

I've attempted to contact the vendor of the phones several times and
have been unsuccessful in reaching anyone.

Anyone had a similar experience?

A snapshot of the phone's config can be seen at
http://www.technologyassociates.ca/phone.jpg 

Thanks,

Ian

Ian Pattison, Senior Analyst
Technology Associates Inc.
Tel: 905-459-2100 ext. 204
Mobile: 416-568-6548
E-mail: [EMAIL PROTECTED] 

 
NOTICE: If received in error, please destroy and notify sender.  Sender
does not waive confidentiality or privilege, and use is prohibited. 
 
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RE: [Asterisk-Users] Grandstream HandyTone-488, * -> FXO problems

[Andrejus Stavickis]

Title: RE: [Asterisk-Users] Grandstream HandyTone-488, * -> FXO problems



Well, the x100p is not always good either. If we forget 
that it only support 600 ohm impedance, the proper example would be the 
problem i have and not being able to overcome is tremendous echo on the VOIP 
phone when i make a call to pstn. after 2 months of trying i had to quit using 
it. 
 
The issue i have is that no matter what i do i never 
receive the output from Asterisk saying somethig else, than "Echo Cancellation: 
0 taps unless TDM bridged, currently OFF" in responce to the command "zap show 
channel 1". this is the ONLY card in the pc, does not share IRQ or IO. It does 
not matter what i put in config files what echo cancellation i use, it just 
never ever goes to something like "currently ON". I've read a lot about echo 
problem on the pstn <-> voip but none of the solution are working for 
me.
Sincerely,--Andyx6722"Outsourcing is akin to 
making a skyscraper taller by taking material from its lower floors."--Byron 
Katz 
 

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Wai 
  WuSent: Wednesday, April 06, 2005 9:47 AMTo: 'Asterisk 
  Users Mailing List - Non-Commercial Discussion'Subject: RE: 
  [Asterisk-Users] Grandstream HandyTone-488, * -> FXO 
  problems
  
  You can stop trying. They still have problem with the firmware 
  concerning the FXO port. If you really want to make a call from * out the 
  PSTN, I suggest you to get a x100p. They are selling it on ebay for $6.99, and 
  I have 4 of those in my * box.
  -Original Message- From: Dan 
  Perik [mailto:[EMAIL PROTECTED]] 
  Sent: Tuesday, April 05, 2005 10:55 PM To: asterisk-users@lists.digium.com Subject: 
  [Asterisk-Users] Grandstream HandyTone-488, * -> FXO problems 
  
  I just got my shiny new Grandstream HandyTone-488 today.  
  My goal is to use it to allow incoming/outgoing calls 
  to PSTN using my normal ole' phone as usual.  I 
  will be switching over to using BroadVoice as my main phone #, but want that to be as seemless of a switchover as 
  possible (for the wife and kids, and for people 
  needing to call us). 
  I've got the following working: 
  FXS -> * ( and then -> BroadVoice ) ( BroadVoice -> ) * -> FXS FXO -> * ( 
  and then -> FXS ) 
  I don't have this working: ( FXS -> 
  ) * -> FXO 
  In other words, I can't seem to call out on my PSTN line from 
  Asterisk. 
  Here's a snippet from sip.conf: [gs1-FXO] type=friend context=default host=dynamic username=gs1-FXO secret= 
  nat=no canreinvite=yes 
  dtmfmode=info incominglimit=1 
  disallow=all allow=ulaw 
  allow=alaw allow=g723.1 
  allow=g729 
  Here's a snippet from extensions.conf: [gs1-fxo-out] exten => 
  _8.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]) 
  So when I dial, say 85429411, I would expect it to dial 
  5429411 out on the PSTN line. I end up not getting any 
  tone or other audio out of the handset.  But, 
  using another phone directly connected to the PSTN, I find that the Grandstream has taken the line off hook, but not 
  dialed any digits.  I get this in my * log when I 
  dial 85429411. 
      -- Executing Dial("SIP/gs1-FXS-9041", 
  "SIP/[EMAIL PROTECTED]") in new stack     -- Called [EMAIL PROTECTED]     -- SIP/gs1-FXO-877b is ringing     -- SIP/gs1-FXO-877b answered SIP/gs1-FXS-9041 
      -- Attempting native bridge of 
  SIP/gs1-FXS-9041 and SIP/gs1-FXO-877b   == Spawn 
  extension (outgoing-ok, 85429411, 1) exited non-zero on 'SIP/gs1-FXS-9041' 
  I know the Handy-Tone 488 is a new device, so there may be 
  some quirks to it.  But I would think it _should_ 
  work. 
  Any suggestions? 
  Thanks! Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users 
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Re: [Asterisk-Users] Cisco 7960 forgets VLAN setting

[Ron Wellsted]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Stefan Gofferje wrote:
> Ron Wellsted schrieb:
> 
>> I have being seeing an interesting problem.  We have and Asterisk
>> 1.0.5 server driving some Cisco 7960s with SIP 7.3 firmware, via a
>> Netgear FSM7326P switch.  We have configured the network to work with
>> data on VLAN 1 and VoIP on VLAN 2.  Every few days, a phone will
>> forget it's VLAN configuration and change themselves to VLAN 0.  As a
>> result they can no longer communicate with the Asterisk server.
>>
>> Has anybody seen any similar problems?
>>
>> If so, what is the solution?
> 
> 
> Do you configure the phones via menu or via TFTP? The Cisco phones tend
> to look for a TFTP server to refresh their config from time to time
> (configurable). I would guess, your problem has something to do with
> that. IF you use TFTP, double check the config files for the VLAN settings.
> A very easy solution would be to use a Cisco switch (2950 is the
> cheapest) as it automatically configures Cisco phones to the right VLAN
> (the voice VLAN configured in the switch).

The VLAN has been configured on the phone, the rest of the config comes
from the tftp server.  I was unable to find any way of setting the vlan
via the tftp server.

- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961  Gossiptel:9309811
N 52.567623, W 2.137621
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.1 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org

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[Asterisk-Users] SIP messages truncated to 256 characters

[Hendrik Magilsen]

 

 

-Original Message-
From: Hendrik Magilsen
[mailto:[EMAIL PROTECTED] 
Sent: Wednesday, April 06, 2005
4:13 PM
To:
asterisk-users@lists.digium.com
Subject: SIP messages truncated to
256 characters

 

I’m troubleshooting a REFER
message.  It appears that incoming REFER message details are being
truncated to 256 characters.  I’m dealing with a SIP provider that
sends out large reference strings.  The r: parameter is just a bit bigger
than the 256 characters asterisk seems to be truncating to, consequently it
loses part of the reference domain and the transaction fails.  I’m
poking through the source code in chan_sip.c, but C is not my strong suit, can
someone point me to the variable declarations for incoming message strings so I
can increase the size and recompile to see if this sorts out my call issue?

 

What is sent

r:


 

What Asterisk sees

Apr  6 15:32:23 WARNING[31877]:
chan_sip.c:995 ditch_braces: No closing brace in '

 

 

Thanks






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[Asterisk-Users] Liveviop problem

[Andrejus Stavickis]

Hi,

I'm just curious if someone had/has a problem with livevoip. When I try
to make an outgoing call, I receive:
-- Called :@217.160.244.186/x037378896
Apr  2 16:47:21 WARNING[10153]: chan_iax2.c:5546 socket_read: Call
rejected by 217.160.244.186: No authority found

The , and first 5 digits of the phone is modified in
this log.

I tried to call Livevoip, they said send us an e-mail and I did, but no
response whatsoever for about a week now.

Sincerely,

--Andy
x6722
 
"Outsourcing is akin to making a skyscraper taller by taking material
from its lower floors."
--Byron Katz 

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[Asterisk-Users] Cisco 7940 Outgoing Audio

[Bellows, Jared]

I’m a Cisco 7940 phone using SCCP.  My setup is a
private network with the * box acting as dhcp server and also tftp
server.  The phone loads and dials out fine.  I can hear the other
person, but there is no outgoing audio.  I’ve read that this is an
RTP problem and have tried making some changes in /etc/hosts to point to my *
box IP but with no luck.  When I do a tcpdump I see that the RTP packets
are sent to 0.0.0.0.  How do I get the phone to send to the * box?

 

Thanks






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Re: [Asterisk-Users] "Choppy" sounds after transferring to ISDN client or after a time

[Michiel van Baak]

On 22:12, Wed 06 Apr 05, Stefan Gofferje wrote:
> Hi folks,
> 
> I'm experiencing a funny problem. I have an Asterisk with a Fritz!PCI 
> card as external and a HFC-S card as internal ISDN. I also use a Cisco 
> SIP phone and X-Lite.
> 
> If I dial out or pick up a call at an ISDN phone, afetr about 30 
> minutes, the other party heavily complains about "choppy" sound, making 
> me hardly understandable.
> If I use a SIP phone and transfer to an ISDN phone, this effect appears 
> immediately.
> If I use only SIP clients, everything is fine.
> 
> I used the monitor application to monitor calls but in the recording, 
> everything is ok. No choppy sound. I guess, this is a codec problem as I 
> once accidentally directly connected a client using ulaw and a client 
> using alaw which resulted in hardly understandable, choppy sound.
> 
> Anyway, is the behavior I described above a bug in Asterisk or is this a 
> config problem?
> Anyone any clue on that?
> 
> Besides, when using an ISDN client, after the same time, the sound goes 
> weird, the channel complains about "sync lost" and I may have some CPU 
> throtteling enabeled.
> I deleted the warning message from the zaphfc channel source as the 
> resulting logging eats up all CPU time on my Athlon XP 2400+ w/ 512MB RAM.
> 
> Regards,
> Stefan

Hi Stefan,

I had the exact same thing when using a cheap HFC-S card
connected to my outside ISDN line. Replacing the card with
an AVM Fritz!PCI fixed this issue for me.
I tried a lot with the HFC-S card, different archs, SMP,
uniprocessor, nolapic, noapic, dual channel ram setup, 1
dimm only, removed all cards but the HFC-S, 4 different
versions of bristuffed. Nothing solved it.

-- 
Michiel van Baak
http://lunteren.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

"Two of the most famous products of Berkeley are LSD and BSD. I don't think 
that this is a coincidence."

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RE: [Asterisk-Users] Cisco 7940 Outgoing Audio

[Alexander Lopez]

Check iptables
with iptables –l 

 

You may be blocking the RTP streams going
INTO your * box.

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bellows, Jared
Sent: Wednesday, April 06, 2005
3:28 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk-Users] Cisco
7940 Outgoing Audio

 

I’m a Cisco 7940 phone using
SCCP.  My setup is a private network with the * box acting as dhcp server
and also tftp server.  The phone loads and dials out fine.  I can
hear the other person, but there is no outgoing audio.  I’ve read
that this is an RTP problem and have tried making some changes in /etc/hosts to
point to my * box IP but with no luck.  When I do a tcpdump I see that the
RTP packets are sent to 0.0.0.0.  How do I get the phone to send to the *
box?

 

Thanks






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Re: [Asterisk-Users] wcte11xp works only after cold reboot

[Rich Adamson]

> > I have also seen this problem on two different asterisk servers using
> > TDM400p cards.
> > 
> > I have not been able to resolve it. If you do an lspci you can see that
> > the system can see the devices but the zaptel drivers don't see them.
> > 
> > I have other systems that work fine and so this has to be down to the
> > combination of motherboard and Digium PCI devices. It would seem that
> > this is a problem with the the Digium hardware because it works after a
> > cold boot.
> > 
> > Does anyone else have a view ?
> > 
> > Stuart
> > 
> > Stuart
> > 
> > On Wed, 2005-04-06 at 12:26 +0200, Alessio Focardi wrote:
> > 
> >>Hi,
> >>
> >>my brand new wcte11xp works like a charme of first boot, then if I
> >>
> >>shutdown -r now
> >>
> >>the server is not detected at reboot ("no such device" after modprobe).
> >>
> >>Turning off the pc and cold restarting fixes the problem.
> >>
> >>Has someone experienced such behaviour before ?
> >>
> >>Tnx for any help!
> 
> This is a known issue and crops up on the list from time to time.  For
> some reason, the card will change is PCI bus ID.  It is set back to the
> correct ID by a cold reboot.
> 
> To work around this, edit the wcte1xxp.c in the zaptel directory so that
> the pci_device_id structure becomes:
> 
> static struct pci_device_id t1xxp_pci_tbl[] = {
> { 0xe159, 0x0001, 0x71fe, PCI_ANY_ID, 0, 0, (unsigned long)
> "Digium Wildcard TE110P T1/E1 Board" },
> { 0xe159, 0x0001, 0x79fe, PCI_ANY_ID, 0, 0, (unsigned long)
> "Digium Wildcard TE110P T1/E1 Board" },
> { 0xe159, 0x0001, 0x795e, PCI_ANY_ID, 0, 0, (unsigned long)
> "Digium Wildcard TE110P T1/E1 Board" },
> { 0xe159, 0x0001, 0x79de, PCI_ANY_ID, 0, 0, (unsigned long)
> "Digium Wildcard TE110P T1/E1 Board" },
> { 0xe159, 0x0001, 0x797e, PCI_ANY_ID, 0, 0, (unsigned long)
> "Digium Wildcard TE110P T1/E1 Board" },
> { 0 }
> };
> 
> This should allow reboots but YMMV (well, it works on my server!).

Are you sure about the above. The OP says he using a TDM400p (fxo/fxs),
not a TE110. Two different animals, I believe.


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RE: [Asterisk-Users] Liveviop problem

[Wiley Siler]

Run this from the CLI...

iax2 show registry

Do you see an entry that matches your LiveVoIP server IP (east or west
coast) and is it registered?

W
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrejus
Stavickis
Sent: Wednesday, April 06, 2005 1:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Liveviop problem

Hi,

I'm just curious if someone had/has a problem with livevoip. When I try
to make an outgoing call, I receive:
-- Called :@217.160.244.186/x037378896
Apr  2 16:47:21 WARNING[10153]: chan_iax2.c:5546 socket_read: Call
rejected by 217.160.244.186: No authority found

The , and first 5 digits of the phone is modified in
this log.

I tried to call Livevoip, they said send us an e-mail and I did, but no
response whatsoever for about a week now.

Sincerely,

--Andy
x6722
 
"Outsourcing is akin to making a skyscraper taller by taking material
from its lower floors."
--Byron Katz 

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Re: [Asterisk-Users] Liveviop problem

[geek]

That is how livevoip deals with problems > /dev/null 


On Wed, 2005-04-06 at 15:23, Andrejus Stavickis wrote:
>   Hi,
> 
> I'm just curious if someone had/has a problem with livevoip. When I try
> to make an outgoing call, I receive:
> -- Called :@217.160.244.186/x037378896
> Apr  2 16:47:21 WARNING[10153]: chan_iax2.c:5546 socket_read: Call
> rejected by 217.160.244.186: No authority found
> 
> The , and first 5 digits of the phone is modified in
> this log.
> 
> I tried to call Livevoip, they said send us an e-mail and I did, but no
> response whatsoever for about a week now.
> 
> Sincerely,
> 
> --Andy
> x6722
>  
> "Outsourcing is akin to making a skyscraper taller by taking material
> from its lower floors."
> --Byron Katz 
> 
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Re: [Asterisk-Users] SIP messages truncated to 256 characters

[Mikael Magnusson]

On Wed, Apr 06, 2005 at 04:17:00PM -0400, Hendrik Magilsen wrote:
> I'm troubleshooting a REFER message.  It appears that incoming REFER message
> details are being truncated to 256 characters.  I'm dealing with a SIP
> provider that sends out large reference strings.  The r: parameter is just a
> bit bigger than the 256 characters asterisk seems to be truncating to,
> consequently it loses part of the reference domain and the transaction
> fails.  I'm poking through the source code in chan_sip.c, but C is not my
> strong suit, can someone point me to the variable declarations for incoming
> message strings so I can increase the size and recompile to see if this
> sorts out my call issue?
>  

Probably one of the tmp variables in get_refer_info.

/Mikael Magnusson

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RE: [Asterisk-Users] wcte11xp works only after cold reboot

[Stuart Hirst]

Stuart Hirst wrote:
> Alessio,
>
> I have also seen this problem on two different asterisk servers using
> TDM400p cards.
>
> I have not been able to resolve it. If you do an lspci you can see that
> the system can see the devices but the zaptel drivers don't see them.
>
> I have other systems that work fine and so this has to be down to the
> combination of motherboard and Digium PCI devices. It would seem that
> this is a problem with the the Digium hardware because it works after a
> cold boot.
>
> Does anyone else have a view ?
>
> Stuart
>
> Stuart
>
> On Wed, 2005-04-06 at 12:26 +0200, Alessio Focardi wrote:
>
>>Hi,
>>
>>my brand new wcte11xp works like a charme of first boot, then if I
>>
>>shutdown -r now
>>
>>the server is not detected at reboot ("no such device" after modprobe).
>>
>>Turning off the pc and cold restarting fixes the problem.
>>
>>Has someone experienced such behaviour before ?
>>
>>Tnx for any help!

>>>This is a known issue and crops up on the list from time to time.  For
>>>some reason, the card will change is PCI bus ID.  It is set back to the
>>>correct ID by a cold reboot.
>>>
>>>To work around this, edit the wcte1xxp.c in the zaptel directory so that
>>>the pci_device_id structure becomes:
>>>
>>>static struct pci_device_id t1xxp_pci_tbl[] = {
>>>{ 0xe159, 0x0001, 0x71fe, PCI_ANY_ID, 0, 0, (unsigned long)
>>>"Digium Wildcard TE110P T1/E1 Board" },
>>>{ 0xe159, 0x0001, 0x79fe, PCI_ANY_ID, 0, 0, (unsigned long)
>>>"Digium Wildcard TE110P T1/E1 Board" },
>>>{ 0xe159, 0x0001, 0x795e, PCI_ANY_ID, 0, 0, (unsigned long)
>>>"Digium Wildcard TE110P T1/E1 Board" },
>>>{ 0xe159, 0x0001, 0x79de, PCI_ANY_ID, 0, 0, (unsigned long)
>>>"Digium Wildcard TE110P T1/E1 Board" },
>>>{ 0xe159, 0x0001, 0x797e, PCI_ANY_ID, 0, 0, (unsigned long)
>>>"Digium Wildcard TE110P T1/E1 Board" },
>>>{ 0 }
>>>};
>>>
>>>This should allow reboots but YMMV (well, it works on my server!).
>>>
>>>HTH
>>>
>>>- --
>>>Ron Wellsted

I don't suppose any one can confirm this type of fix for the TDM boards. I
can make a guess but the systems are live and I will need to schedule outage
to attempt this fix.

Thanks,

Stuart
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.9.4 - Release Date: 06/04/2005

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Re: [Asterisk-Users] Cisco 7940 Outgoing Audio

[Andy Hamilton]

Which skinny driver are you using?

Also, what about incoming calls?

Post your skinny.conf or sccp.conf as well as the config file for your
7940 from the TFTP server. It may help also if you include any other
generic Cisco config files from your TFTP directory, for example an
OS79XX.TXT.

-Andy


On Apr 6, 2005 3:27 PM, Bellows, Jared <[EMAIL PROTECTED]> wrote:
>  
>  
> 
> I'm a Cisco 7940 phone using SCCP.  My setup is a private network with the *
> box acting as dhcp server and also tftp server.  The phone loads and dials
> out fine.  I can hear the other person, but there is no outgoing audio. 
> I've read that this is an RTP problem and have tried making some changes in
> /etc/hosts to point to my * box IP but with no luck.  When I do a tcpdump I
> see that the RTP packets are sent to 0.0.0.0.  How do I get the phone to
> send to the * box? 
> 
>   
> 
> Thanks 
> ___
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>   
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[Asterisk-Users] AMP & Handset Provisioning

[Stuart Hirst]

I have in the past written a hack to AMP 1.10.004 to automatically generate
Polycom configuration files such that when an extension is created you can
select the type of handset as a Polycom and then enter the handsets MAC /
Serial on the page and AMP would automatically create the correctly
formatted config files. This was a dirty hack written in haste.

I am about to start doing a patch for the latest version of AMP that will
allow developers to added their own handset config plug-in or at least
publish the method used to recreate correctly the previous Polycom hack such
that anyone should be able to build their own handset configuration
templates and have AMP auto provision whilst maintaining some consistency.

Has anyone else done anything similar given that duplication is the root of
all evil ?

Regards,

Stuart Hirst

--
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Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.9.4 - Release Date: 06/04/2005

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Re: [Asterisk-Users] Liveviop problem

[Rich Adamson]

> I'm just curious if someone had/has a problem with livevoip. When I try
> to make an outgoing call, I receive:
> -- Called :@217.160.244.186/x037378896
> Apr  2 16:47:21 WARNING[10153]: chan_iax2.c:5546 socket_read: Call
> rejected by 217.160.244.186: No authority found
> 
> The , and first 5 digits of the phone is modified in
> this log.
> 
> I tried to call Livevoip, they said send us an e-mail and I did, but no
> response whatsoever for about a week now.

Been working like a charm here. Just got off the phone.

The few times I've had to contact support via email, I had a response
with 15 minutes. I might have caught them on a rather slow day though.


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RE: [Asterisk-Users] Liveviop problem

[Wiley Siler]

I have had far better luck than that too.  More like a hour for me but
that is not too bad.

W
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Wednesday, April 06, 2005 3:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Liveviop problem


> I'm just curious if someone had/has a problem with livevoip. When I 
> try to make an outgoing call, I receive:
> -- Called :@217.160.244.186/x037378896
> Apr  2 16:47:21 WARNING[10153]: chan_iax2.c:5546 socket_read: Call 
> rejected by 217.160.244.186: No authority found
> 
> The , and first 5 digits of the phone is modified in

> this log.
> 
> I tried to call Livevoip, they said send us an e-mail and I did, but 
> no response whatsoever for about a week now.

Been working like a charm here. Just got off the phone.

The few times I've had to contact support via email, I had a response
with 15 minutes. I might have caught them on a rather slow day though.


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Re: [Asterisk-Users] wcte11xp works only after cold reboot

[Ron Wellsted]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Stuart Hirst wrote:
> 
> Stuart Hirst wrote:
> 
>>Alessio,
>>
>>I have also seen this problem on two different asterisk servers using
>>TDM400p cards.
>>
>>I have not been able to resolve it. If you do an lspci you can see that
>>the system can see the devices but the zaptel drivers don't see them.
>>
>>I have other systems that work fine and so this has to be down to the
>>combination of motherboard and Digium PCI devices. It would seem that
>>this is a problem with the the Digium hardware because it works after a
>>cold boot.
>>
>>Does anyone else have a view ?
>>
>>Stuart
>>
>>Stuart
>>
>>On Wed, 2005-04-06 at 12:26 +0200, Alessio Focardi wrote:
>>
>>
>>>Hi,
>>>
>>>my brand new wcte11xp works like a charme of first boot, then if I
>>>
>>>shutdown -r now
>>>
>>>the server is not detected at reboot ("no such device" after modprobe).
>>>
>>>Turning off the pc and cold restarting fixes the problem.
>>>
>>>Has someone experienced such behaviour before ?
>>>
>>>Tnx for any help!
> 
> 
This is a known issue and crops up on the list from time to time.  For
some reason, the card will change is PCI bus ID.  It is set back to the
correct ID by a cold reboot.

To work around this, edit the wcte1xxp.c in the zaptel directory so that
the pci_device_id structure becomes:

static struct pci_device_id t1xxp_pci_tbl[] = {
   { 0xe159, 0x0001, 0x71fe, PCI_ANY_ID, 0, 0, (unsigned long)
"Digium Wildcard TE110P T1/E1 Board" },
   { 0xe159, 0x0001, 0x79fe, PCI_ANY_ID, 0, 0, (unsigned long)
"Digium Wildcard TE110P T1/E1 Board" },
   { 0xe159, 0x0001, 0x795e, PCI_ANY_ID, 0, 0, (unsigned long)
"Digium Wildcard TE110P T1/E1 Board" },
   { 0xe159, 0x0001, 0x79de, PCI_ANY_ID, 0, 0, (unsigned long)
"Digium Wildcard TE110P T1/E1 Board" },
   { 0xe159, 0x0001, 0x797e, PCI_ANY_ID, 0, 0, (unsigned long)
"Digium Wildcard TE110P T1/E1 Board" },
   { 0 }
};

This should allow reboots but YMMV (well, it works on my server!).

HTH

- --
Ron Wellsted
> 
> 
> I don't suppose any one can confirm this type of fix for the TDM boards. I
> can make a guess but the systems are live and I will need to schedule outage
> to attempt this fix.
> 
> Thanks,
> 
> Stuart

My mistake, the above fix is for the TE110P, however I seem to recall a
similar problem being reported on the TDM board,certainly the symptom
and the cure is the same.  You can try using "lspci -vn" to check the
pci and subsytem IDs being returned by your card and check that these
match the values in the relevant source code file.  If not try adding
the relevant entries to the pci_device_id table.

- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961  Gossiptel:9309811
N 52.567623, W 2.137621
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Re: [Asterisk-Users] AMP & Handset Provisioning

[Jason Becker]

Hi Stuart,
Stuart Hirst wrote:
I have in the past written a hack to AMP 1.10.004 to automatically generate
Polycom configuration files such that when an extension is created you can
select the type of handset as a Polycom and then enter the handsets MAC /
Serial on the page and AMP would automatically create the correctly
formatted config files. This was a dirty hack written in haste.
I am about to start doing a patch for the latest version of AMP that will
allow developers to added their own handset config plug-in or at least
publish the method used to recreate correctly the previous Polycom hack such
that anyone should be able to build their own handset configuration
templates and have AMP auto provision whilst maintaining some consistency.
Has anyone else done anything similar given that duplication is the root of
all evil ?
 

A Feature Request for this very capability was submitted earlier today:
http://sourceforge.net/tracker/index.php?func=detail&aid=1178008&group_id=121515&atid=690575
We'd welcome your contribution and can work with you to add this 
valuable feature to AMP.

Regards,
--
Jason Becker
Director & CEO
Coalescent Systems Inc.
Enabling Open Source Telephony
403.244.8089
www.coalescentsystems.ca
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[Asterisk-Users] How to avoid that certain calls come into the voicemail (e.g. wakeup calls)?

[Ronald Wiplinger]

We use wakeup calls for reminders, but it happens, that the person to be 
reminded is on the phone. To get a voicemail later is not really useful 
anymore, ...
Is there a way to avoid that?

bye
Ronald
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[Asterisk-Users] Context overlap?

[Dylan VanHerpen]

I have an auto-attendant for day and night. When the [businesshours]
AA runs, it executes exten => s1 through exten => s9, then continues
with exten => s10 in [nightmode], even though they are in different
contexts. This was working fine until I added more 's' extensions in
night mode. When I comment out all 's' extensions 10 and above in
[nightmode], it works fine again.

Is this a bug or am I missing something?

Dylan.

Asterisk CVS-v1-0-02/24/05-13:18:42


[auto-attendant]
; Business Hours
include => businesshours|08:00-16:59|mon-fri|*|*

; After Hours (anything that doesn't match holiday or businesshours)
include => nightmode

[businesshours]
exten => s,1,Answer
exten => s,2,DigitTimeout,15
exten => s,3,ResponseTimeout,20
exten => s,4,Background(thank-you-for-calling)
exten => s,5,Background(if-u-know-ext-dial)
...
exten => s,9,Background(to-hear-menu-again)

[nightmode]
exten => s,1,Answer
exten => s,2,DigitTimeout,15
exten => s,3,ResponseTimeout,20
exten => s,4,Background(thank-you-for-calling)
exten => s,5,Background(if-u-know-ext-dial)
...
exten => s,10,Background()
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Re: [Asterisk-Users] How to avoid that certain calls come into the voicemail (e.g. wakeup calls)?

[Andy Hamilton]

Ronald:

Try reducing the duration for which the wakeup call waits on the
person it is calling; my default is 20 seconds.
Here is a sample of one of my wakeup call files, automatically generated:

Channel: SIP/500
MaxRetries: 1
RetryTime: 10
WaitTime: 20
Application: MusicOnHold
Callerid: * Wakeup Call

Twenty seconds works for me (without the call going to VM), but I
tried setting it to 200. Asterisk said is was an invalid time and the
call timed out and hungup after about 40sec.
It also looks to me, on the test I just ran, that it will try calling
twice, even though MaxRetries is set to 1.

-Andy

On Apr 6, 2005 5:30 PM, Ronald Wiplinger <[EMAIL PROTECTED]> wrote:
> We use wakeup calls for reminders, but it happens, that the person to be
> reminded is on the phone. To get a voicemail later is not really useful
> anymore, ...
> Is there a way to avoid that?
> 
> bye
> 
> Ronald
> 
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[Asterisk-Users] Cisco 7940 SIP and No compatible codecs!

[Ron Joffe]

Hey folks,

I have cisco 7940 with SIP configured on Asterisk stable. I can place calls to 
it from either my Snom Sip phone on analog extensions on a zaptel card. But I 
don't seem to be able to place calls from it to either the snom, or the Zap 
extensions.

My log shows:

NOTICE[8923]: chan_sip.c:2894 process_sdp: No compatible codecs!

I can provide more detailed debug if required, but anyone ran across this.

Thanks,

Ron

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Re: [Asterisk-Users] Context overlap?

[Moises Silva]

, if hopefully im understanding your question. "Even tough they
are in different context".. i think thats wrong, since you are
including both contexts in auto-attendant, they both are in the same
context. The bussinesshours run first because its included first (in
the hours set in the configuration), and if you dont have higher S
extensions in the night context all works fine, but when you have more
that 10 (more than bussinesshours) in night, then it continues with
the next S extension. Hope i have been clear, and hope it helps you.

Best Regards

-Moisés Silva

On Apr 6, 2005 10:36 PM, Dylan VanHerpen <[EMAIL PROTECTED]> wrote:
> I have an auto-attendant for day and night. When the [businesshours]
> AA runs, it executes exten => s1 through exten => s9, then continues
> with exten => s10 in [nightmode], even though they are in different
> contexts. This was working fine until I added more 's' extensions in
> night mode. When I comment out all 's' extensions 10 and above in
> [nightmode], it works fine again.
> 
> Is this a bug or am I missing something?
> 
> Dylan.
> 
> Asterisk CVS-v1-0-02/24/05-13:18:42
> 
> [auto-attendant]
> ; Business Hours
> include => businesshours|08:00-16:59|mon-fri|*|*
> 
> ; After Hours (anything that doesn't match holiday or businesshours)
> include => nightmode
> 
> [businesshours]
> exten => s,1,Answer
> exten => s,2,DigitTimeout,15
> exten => s,3,ResponseTimeout,20
> exten => s,4,Background(thank-you-for-calling)
> exten => s,5,Background(if-u-know-ext-dial)
> ...
> exten => s,9,Background(to-hear-menu-again)
> 
> [nightmode]
> exten => s,1,Answer
> exten => s,2,DigitTimeout,15
> exten => s,3,ResponseTimeout,20
> exten => s,4,Background(thank-you-for-calling)
> exten => s,5,Background(if-u-know-ext-dial)
> ...
> exten => s,10,Background()
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[Asterisk-Users] Realtime UPDATE

[Rod Bacon]

I'm sorry if this has been answered before, but I've been through all 
the (lengthy) threads on Realtime, and can't find the answer.

My problem is that upon registration, the UA's IP address and port 
information isn't being written to the MYSQL realtime database. 
Subsequently, calls to the UA fail if they originate from another * 
server (The server DOES attempt a lookup, but obviously gets no value 
for IP address / PORT).

From the MYSQL logs, I see the folowing at registration;
UPDATE sip SET ipaddr = '', port = '', regseconds = '0', username = 
'9998' WHERE name = '9998'

The weird thing is that it was working at some point yesterday.
Can anyone suggest a place to start looking?
Also, how do I enable debug logging so I can see the realtime info in 
the * CLI or logs?

--
==
Rod Bacon - VOIP Systems Engineer
Empowered Communications
Ground Floor, 102 York St. South Melbourne
Victoria, Australia. 3205
Phone: +613 99401600Fax: +613 99401650
==
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Re: [Asterisk-Users] Realtime UPDATE

[Rod Bacon]

I found the logging stuff.
Here is the * debug info.
Apr  7 09:09:20 DEBUG[3672]: res_config_mysql.c:117 realtime_mysql: MySQL 
RealTime: Retrieve SQL: SELECT * FROM sip WHERE name = ''
Apr  7 09:09:20 DEBUG[3672]: res_config_mysql.c:605 mysql_reconnect: MySQL 
RealTime: Everything is fine.
   -- SIP Seeding peers from Astdb: '' at [EMAIL PROTECTED]:5060 for 
3600
Apr  7 09:09:20 DEBUG[3672]: res_config_mysql.c:117 realtime_mysql: MySQL 
RealTime: Retrieve SQL: SELECT * FROM sip WHERE name = ''
Apr  7 09:09:20 DEBUG[3672]: res_config_mysql.c:605 mysql_reconnect: MySQL 
RealTime: Everything is fine.
   -- SIP Seeding peers from Astdb: '' at [EMAIL PROTECTED]:5060 for 
3600
   -- Unregistered SIP ''
Apr  7 09:09:20 DEBUG[3672]: res_config_mysql.c:311 update_mysql: MySQL 
RealTime: Update SQL: UPDATE sip SET ipaddr = '', port = '', regseconds = 
'0', username = '' WHERE name = ''
Apr  7 09:09:20 DEBUG[3672]: res_config_mysql.c:605 mysql_reconnect: MySQL 
RealTime: Everything is fine.
Apr  7 09:09:20 DEBUG[3672]: res_config_mysql.c:330 update_mysql: MySQL 
RealTime: Updated 0 rows on table: sip
Apr  7 09:09:20 DEBUG[3672]: res_config_mysql.c:117 realtime_mysql: MySQL 
RealTime: Retrieve SQL: SELECT * FROM sip WHERE name = ''
Apr  7 09:09:20 DEBUG[3672]: res_config_mysql.c:605 mysql_reconnect: MySQL 
RealTime: Everything is fine.
Apr  7 09:09:20 DEBUG[3672]: db.c:177 ast_db_get: Unable to find key '' 
in family 'SIP/Registry'
Apr  7 09:09:20 DEBUG[3672]: res_config_mysql.c:117 realtime_mysql: MySQL 
RealTime: Retrieve SQL: SELECT * FROM sip WHERE name = ''
Apr  7 09:09:20 DEBUG[3672]: res_config_mysql.c:605 mysql_reconnect: MySQL 
RealTime: Everything is fine.
Apr  7 09:09:20 DEBUG[3672]: db.c:177 ast_db_get: Unable to find key '' 
in family 'SIP/Registry'
   -- Registered SIP '' at 192.168.0.137 port 5060 expires 3600
   -- Saved useragent "Grandstream BT100 1.0.5.22" for peer 
Apr  7 09:09:20 DEBUG[3672]: res_config_mysql.c:311 update_mysql: MySQL 
RealTime: Update SQL: UPDATE sip SET ipaddr = '', port = '', regseconds = 
'0', username = '' WHERE name = ''
Apr  7 09:09:20 DEBUG[3672]: res_config_mysql.c:605 mysql_reconnect: MySQL 
RealTime: Everything is fine.
Apr  7 09:09:20 DEBUG[3672]: res_config_mysql.c:330 update_mysql: MySQL 
RealTime: Updated 0 rows on table: sip
Apr  7 09:09:20 DEBUG[3672]: res_config_mysql.c:117 realtime_mysql: MySQL 
RealTime: Retrieve SQL: SELECT * FROM sip WHERE name = ''
Apr  7 09:09:20 DEBUG[3672]: res_config_mysql.c:605 mysql_reconnect: MySQL 
RealTime: Everything is fine.
   -- SIP Seeding peers from Astdb: '' at [EMAIL PROTECTED]:5060 for 
3600
Apr  7 09:09:35 DEBUG[3672]: chan_sip.c:937 __sip_autodestruct: Auto 
destroying call '[EMAIL PROTECTED]'


- Original Message - 
From: "Rod Bacon" <[EMAIL PROTECTED]>
To: 
Sent: Thursday, April 07, 2005 9:06 AM
Subject: [Asterisk-Users] Realtime UPDATE


I'm sorry if this has been answered before, but I've been through all the 
(lengthy) threads on Realtime, and can't find the answer.

My problem is that upon registration, the UA's IP address and port 
information isn't being written to the MYSQL realtime database. 
Subsequently, calls to the UA fail if they originate from another * server 
(The server DOES attempt a lookup, but obviously gets no value for IP 
address / PORT).

From the MYSQL logs, I see the folowing at registration;
UPDATE sip SET ipaddr = '', port = '', regseconds = '0', username = '9998' 
WHERE name = '9998'

The weird thing is that it was working at some point yesterday.
Can anyone suggest a place to start looking?
Also, how do I enable debug logging so I can see the realtime info in the 
* CLI or logs?

--
==
Rod Bacon - VOIP Systems Engineer
Empowered Communications
Ground Floor, 102 York St. South Melbourne
Victoria, Australia. 3205
Phone: +613 99401600Fax: +613 99401650
==
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[Asterisk-Users] DIDs in 510, 408, 916 415 area code

[M. Ehsanul Karim]

I am looking for DIDs in this area codes for wholesale 


510
408
916
415


Please quote me on pricing for blocks of 10,20,50,100

Thanks.

Ehsanul Karim
[EMAIL PROTECTED]
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RE : [Asterisk-Users] how can i connect a cost display on asterisk

[Hakem Taourchi]

Hello,


Do
you confirm there is a way to send information and update it while the call is
ongoing using the caller Id information ? 

 

 

-Message
d'origine-
De :
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Steve Totaro
Envoyé : mercredi 6 avril
2005 15:48
À : Asterisk Users Mailing
List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users]
how can i connect a cost display on asterisk

 



i
did something simlar by overwriting incoming caller id with other information.







-
Original Message - 





From: Han van Hulst 





To: asterisk-users@lists.digium.com 





Sent: Wednesday,
 April 06, 2005 9:21 AM





Subject:
[Asterisk-Users] how can i connect a cost display on asterisk





 



Is
it possible to connect a display that shows the costs of a call in progress?

 

Can
I also send the call cost to a grandstream display?

 

Greeting

 

Johannes

 







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RE: [Asterisk-Users] Realtime UPDATE

[Thierry Wehr]

> -Message d'origine-
> De : [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] De la part 
> de Rod Bacon
> Envoyé : jeudi 7 avril 2005 01:06
> À : asterisk-users@lists.digium.com
> Objet : [Asterisk-Users] Realtime UPDATE

> My problem is that upon registration, the UA's IP address and 
> port information isn't being written to the MYSQL realtime database. 
> Subsequently, calls to the UA fail if they originate from 
> another * server (The server DOES attempt a lookup, but 
> obviously gets no value for IP address / PORT).

Hi

I'd the same problem and discovered that you have to put

rtnoupdate=no in you'r sip.conf

Hope it helps you

Thierry

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Re: [Asterisk-Users] DIDs in 510, 408, 916 415 area code

[Vern Norman]

Is this a * specific question, or has the intent of this list changed? 
just curious  :)

Try calling the Telco providers in those areas... I doubt that you will find 
them here. most of them don't really like *

- Original Message - 
From: "M. Ehsanul Karim" <[EMAIL PROTECTED]>
To: 
Sent: Wednesday, April 06, 2005 7:12 PM
Subject: [Asterisk-Users] DIDs in 510, 408, 916 415 area code


I am looking for DIDs in this area codes for wholesale
510
408
916
415
Please quote me on pricing for blocks of 10,20,50,100
Thanks.
Ehsanul Karim
[EMAIL PROTECTED]
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Re: [Asterisk-Users] Realtime UPDATE

[Rod Bacon]

I had discovered this myself. Once I set this value, the updates started to 
occur, but as shown in my earlier post, are all NULL values.


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

Sent: Thursday, April 07, 2005 9:36 AM
Subject: RE: [Asterisk-Users] Realtime UPDATE


-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part
de Rod Bacon
Envoyé : jeudi 7 avril 2005 01:06
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] Realtime UPDATE

My problem is that upon registration, the UA's IP address and
port information isn't being written to the MYSQL realtime database.
Subsequently, calls to the UA fail if they originate from
another * server (The server DOES attempt a lookup, but
obviously gets no value for IP address / PORT).
Hi
I'd the same problem and discovered that you have to put
rtnoupdate=no in you'r sip.conf
Hope it helps you
Thierry
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Re: [Asterisk-Users] How to avoid that certain calls come into thevoicemail (e.g. wakeup calls)?

[Rod Bacon]

It also looks to me, on the test I just ran, that it will try calling
twice, even though MaxRetries is set to 1.
That's right. If you only want it to call once, set max REtries to 0.

-Andy
On Apr 6, 2005 5:30 PM, Ronald Wiplinger <[EMAIL PROTECTED]> wrote:
We use wakeup calls for reminders, but it happens, that the person to be
reminded is on the phone. To get a voicemail later is not really useful
anymore, ...
Is there a way to avoid that?
bye
Ronald
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[Asterisk-Users] rookie question

[Randy Paries]

Hello,

Just want to make sure I can do what I need to do with asterisk

I have a very small company most of us work remotely and one locally.

So I have 3 lines, and then 3 Digium Wildcard X100P Cards

So I have each of the lines plugged into the x100's 

Can I use one of the Digium Wildcard X100P's as a local extension as well?
Can asterisk take an incoming call and transfer it to a remote number?

Thanks for the help and patience for these rookies questions

Randy


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[Asterisk-Users] Polycom IP 600 not ringing

[Alex]

Hi guys,
Has anyone come across a problem when Polycom IP 600 does make an audible  
ring sound, even though the call comes in? I can see it on LCD and red  
light flashes. When I pickup the phone, everything is fine. It only  
applies to SIP calls. If the call comes in from PSTN via TDM400 card,  
everything seems to be ok.

The same scenario happens when I try to ring from IP600 other Polycom  
phones, again they don't ring.
I am using 2.6.1 bootrom and 1.4.1 SIP firmware.

Anyone can point me in the right direction?
Thanks,
Alex.
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Re: [Asterisk-Users] Polycom IP 600 not ringing

[Alex]

Sorry...
It should reed Polycom IP 600 does not make an  audible ring sound
...half a sleep :-)
On Thu, 07 Apr 2005 10:07:57 +1000, Alex <[EMAIL PROTECTED]> wrote:
Hi guys,
Has anyone come across a problem when Polycom IP 600 does not make an  
audible ring sound, even though the call comes in? I can see it on LCD  
and red light flashes. When I pickup the phone, everything is fine. It  
only applies to SIP calls. If the call comes in from PSTN via TDM400  
card, everything seems to be ok.

The same scenario happens when I try to ring from IP600 other Polycom  
phones, again they don't ring.
I am using 2.6.1 bootrom and 1.4.1 SIP firmware.

Anyone can point me in the right direction?
Thanks,
Alex.
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Re: [Asterisk-Users] Context overlap?

[Dylan VanHerpen]

Thanks Moisés, that's kindof what I figured. I'll fix it by changing
the nightmode include to an explicit timeframe, instead of the 'catch
all' solution:

; After Hours (anything that doesn't match holiday or businesshours)
include => nightmode|00:00-7:59|mon-fri|*|*
include => nightmode|17:00-23:59|mon-fri|*|*
include => nightmode|*|sat|*|*
include => nightmode|*|sun|*|*

Dylan.

On Apr 6, 2005 5:05 PM, Moises Silva <[EMAIL PROTECTED]> wrote:
> , if hopefully im understanding your question. "Even tough they
> are in different context".. i think thats wrong, since you are
> including both contexts in auto-attendant, they both are in the same
> context. The bussinesshours run first because its included first (in
> the hours set in the configuration), and if you dont have higher S
> extensions in the night context all works fine, but when you have more
> that 10 (more than bussinesshours) in night, then it continues with
> the next S extension. Hope i have been clear, and hope it helps you.
> 
> Best Regards
> 
> -Moisés Silva
> 
> On Apr 6, 2005 10:36 PM, Dylan VanHerpen <[EMAIL PROTECTED]> wrote:
> > I have an auto-attendant for day and night. When the [businesshours]
> > AA runs, it executes exten => s1 through exten => s9, then continues
> > with exten => s10 in [nightmode], even though they are in different
> > contexts. This was working fine until I added more 's' extensions in
> > night mode. When I comment out all 's' extensions 10 and above in
> > [nightmode], it works fine again.
> >
> > Is this a bug or am I missing something?
> >
> > Dylan.
> >
> > Asterisk CVS-v1-0-02/24/05-13:18:42
> >
> > [auto-attendant]
> > ; Business Hours
> > include => businesshours|08:00-16:59|mon-fri|*|*
> >
> > ; After Hours (anything that doesn't match holiday or businesshours)
> > include => nightmode
> >
> > [businesshours]
> > exten => s,1,Answer
> > exten => s,2,DigitTimeout,15
> > exten => s,3,ResponseTimeout,20
> > exten => s,4,Background(thank-you-for-calling)
> > exten => s,5,Background(if-u-know-ext-dial)
> > ...
> > exten => s,9,Background(to-hear-menu-again)
> >
> > [nightmode]
> > exten => s,1,Answer
> > exten => s,2,DigitTimeout,15
> > exten => s,3,ResponseTimeout,20
> > exten => s,4,Background(thank-you-for-calling)
> > exten => s,5,Background(if-u-know-ext-dial)
> > ...
> > exten => s,10,Background()
> > ___
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> > http://lists.digium.com/mailman/listinfo/asterisk-users
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>
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Re: [Asterisk-Users] No more updates of IP address and port in CVS HEAD

[Rod Bacon]

Bugger...
That explains it...

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

Sent: Sunday, March 20, 2005 2:08 AM
Subject: [Asterisk-Users] No more updates of IP address and port in CVS HEAD


Good afternoon
Since the cvs version of yesterday, the ip address and the port of the
sipfriend are no more updated in the realtime database
Regards
Thierry Wehr
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Re: [Asterisk-Users] Web interface for realtime Mysql friends/peer

[Matt Gibson]

Kanuri, Seshu (Company IT) wrote:
Marshall,
I am interested in seeing what you wrote to manage MySQL database
objects. 

By the way, latest version of OpenOffice comes with a MySQL
Administrator GUI to manage tables and data. This is something to look
at too.
Seshu Kanuri

 I am also interested in seeing what you've made up too :)
Thanks,
Matt
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[Asterisk-Users] Asterisk and broadsoft

[Digital Support Technologies]

Hello,

 

I have a asterisk box that is connected to a broadsoft commercial
switch.  I am using asterisk with its queue application for a multitude of
things.  My new carrier requires that I send my ANI over when placing a
outbound call to authorize me.  My cell phone is not in that list of allowed
numbers.  What I want to know if there is a way around this and still allowing
asterisk to pass CID info over to the cell phone so I can see who is calling

 

Thanks









avast! Antivirus: Outbound message clean.
Virus Database (VPS): 0514-0, 04/05/2005Tested on: 4/6/2005 8:30:01 PMavast! - copyright (c) 1988-2005 ALWIL Software.




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RE: [Asterisk-Users] DIDs in 510, 408, 916 415 area code

[Digital Support Technologies]

We can do DID's in those area's cost will be around 10.00 a number 

-Original Message- 
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vern Norman 
Sent: Wednesday, April 06, 2005 7:37 PM 
To: M. Ehsanul Karim; Asterisk Users Mailing List - Non-Commercial
Discussion 
Subject: Re: [Asterisk-Users] DIDs in 510, 408, 916 415 area code 

Is this a * specific question, or has the intent of this list changed? 
just curious  :) 


Try calling the Telco providers in those areas... I doubt that you will find

them here. most of them don't really like * 

- Original Message - 
From: "M. Ehsanul Karim" <[EMAIL PROTECTED]> 
To:  
Sent: Wednesday, April 06, 2005 7:12 PM 
Subject: [Asterisk-Users] DIDs in 510, 408, 916 415 area code 


>I am looking for DIDs in this area codes for wholesale 
> 
> 
> 510 
> 408 
> 916 
> 415 
> 
> 
> Please quote me on pricing for blocks of 10,20,50,100 
> 
> Thanks. 
> 
> Ehsanul Karim 
> [EMAIL PROTECTED] 
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RE: [Asterisk-Users] SIP messages truncated to 256 characters

[Hendrik A. Magilsen]

That did it, the error went away, but the solution doesn't solve the
problem.  I probably will start a new thread when I get more data because
it's a bit convoluted.  I'm using Asterisk to act as an intermediary between
two sip providers (one provides DID's to users on the other, with Asterisk
acting more or less like a back-to-back agent in the middle).  However when
the user on the SIP side tries to redirect to a conference server the call
breaks down in the asterisk. Asterisk seems to see the refer message as a
new call and doesn't relate it to the original call leg.

Like I said convoluted, and my approach to the problem may not be the right
one.

Thanks for the help
Hendrik


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mikael
Magnusson
Sent: Wednesday, April 06, 2005 4:48 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] SIP messages truncated to 256 characters

On Wed, Apr 06, 2005 at 04:17:00PM -0400, Hendrik Magilsen wrote:
> I'm troubleshooting a REFER message.  It appears that incoming REFER
message
> details are being truncated to 256 characters.  I'm dealing with a SIP
> provider that sends out large reference strings.  The r: parameter is just
a
> bit bigger than the 256 characters asterisk seems to be truncating to,
> consequently it loses part of the reference domain and the transaction
> fails.  I'm poking through the source code in chan_sip.c, but C is not my
> strong suit, can someone point me to the variable declarations for
incoming
> message strings so I can increase the size and recompile to see if this
> sorts out my call issue?
>  

Probably one of the tmp variables in get_refer_info.

/Mikael Magnusson

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Re: [Asterisk-Users] SIP - SIP Problems

[Joseph]

On Wed, 2005-04-06 at 14:49 -0400, Ian Pattison wrote:
> Hi Everybody...
> 
> Continuing the litany of problems I'm experiencing with my new system I'm 
> getting issues connecting between SIP phones.
> 
> A bit of background... I have an asterisk server running in a central 
> location where I have two incoming analog lines connected to FXO ports, two 
> analog phones connecting to FXS ports and a single SIP phone. In addition I 
> have a remote site connected via a CIPE VPN (ok..ok I know it's not a real 
> VPN...) with a single SIP phone. 
> 
> Calls initiated from the remote SIP phone to the central SIP phone often have 
> trouble... the user of the central phone cannot hear anything from the remote 
> phone although everything is heard at the remote phone. If the remote phone 
> calls either outside or to one of the Zap phones there is no trouble. If the 
> local SIP phone calls the remote SIP phone there is no trouble. Both phones 
> are from the same vendor, have the same firmware and the same configuration 
> with the exception of phone number, PIN, IP address etc.
> 
> What am I doing wrong here?
> 
> Ian

What *-ver. are you using?
I had a problem with SIP in ver.1.0.7 and 1.0.6 so I'm still on 1.0.5

-- 
#Joseph
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Re: [Asterisk-Users] SIP - SIP Problems

[Rod Bacon]

I'd personally be using Ethereal to look inside the SIP messages for the SDP 
info and checking the source/destination of the resultant RTP stream. 
One-way audio is typical of NAT issues. Although you are running a VPN (of 
sorts) I suspect that your SDP messages are getting screwed up somewhere.

What are the asterisk NAT settings in effect for each of the SIP phones? I'd 
be inclined to turn them both ON to ensure that symmetrical RTP in being 
used. Also make sure that canreinvite is OFF for both.

- Original Message - 
From: "Ian Pattison" <[EMAIL PROTECTED]>
To: 
Sent: Thursday, April 07, 2005 4:49 AM
Subject: [Asterisk-Users] SIP - SIP Problems

Hi Everybody...
Continuing the litany of problems I'm experiencing with my new system I'm 
=etting issues connecting between SIP phones.

A bit of background... I have an asterisk server running in a central 
=ocation where I have two incoming analog lines connected to FXO ports, =wo 
analog phones connecting to FXS ports and a single SIP phone. In =ddition I 
have a remote site connected via a CIPE VPN (ok..ok I know it's =ot a real 
VPN...) with a single SIP phone.

Calls initiated from the remote SIP phone to the central SIP phone often 
=ave trouble... the user of the central phone cannot hear anything from =he 
remote phone although everything is heard at the remote phone. If the =emote 
phone calls either outside or to one of the Zap phones there is no =rouble. 
If the local SIP phone calls the remote SIP phone there is no =rouble. Both 
phones are from the same vendor, have the same firmware and =he same 
configuration with the exception of phone number, PIN, IP address =tc.

What am I doing wrong here?
Ian
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Re: RE : [Asterisk-Users] how can i connect a cost display on asterisk

[Eric Wieling]

Hakem Taourchi wrote:
Hello, 
Do you confirm there is a way to send information and update it while
the call is ongoing using the caller Id information ? 
I strongly doubt this will work on anything except an analog phone.  I 
also strongly doubt that Asterisk supports this at all.

--
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Mark Twain
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[Asterisk-Users] Asterisk on Slack 10.0

[Gary Guthary]

Hello -

I'm (VERY) new to the mailing list.

And after seeing some of the postings here, I can also see that I'm also
VERY new to the Asterisk game. - WOW!  I can see this'll take a while to
'catch up' (if that ever happens).

You folks have enough room for one more to play?

Without boring a lot of you & trying to practice a little -
'user-list-etiquitte', I'll keep this short.

I just got my dev-kit & am trying to get Asterisk compiled & running on a
Slack-10.0 box (2.4.26 kernel). - Reason is that I have 7-8 yrs experience
on Slack & little with R/H & none with Fedora. - Besides, the docs with
'cutesy' little CD that came with the dev-kit don't match up with the Fedora
release levels that are on line today. - Decided to give up & go back to
slack-10.0 (since 2.4.26 is 'somewhat' supported).

If there's anybody out there that's done something similar on a slack box,
I'd like to talk to you.

If I'm posting in the wrong place, I'm sorry. - Could somebody please steer
me in the right direction.

Thanks in advance.

Gary


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Re: [Asterisk-Users] Polycom IP 600 not ringing

[Eric Wieling]

Alex wrote:
Sorry...
It should reed Polycom IP 600 does not make an  audible ring sound
...half a sleep :-)
On Thu, 07 Apr 2005 10:07:57 +1000, Alex <[EMAIL PROTECTED]> wrote:
Hi guys,
Has anyone come across a problem when Polycom IP 600 does not make an  
audible ring sound, even though the call comes in? I can see it on 
LCD  and red light flashes. When I pickup the phone, everything is 
fine. It  only applies to SIP calls. If the call comes in from PSTN 
via TDM400  card, everything seems to be ok.

The same scenario happens when I try to ring from IP600 other Polycom  
phones, again they don't ring.
I am using 2.6.1 bootrom and 1.4.1 SIP firmware.

Anyone can point me in the right direction?
I have seen this problem on the Polycom 500.  ONLY happens when I call 
from port 1 of my SPA-2000, works fine on port 2.  I cannot see any 
significant difference in the way the two ports are configured (on the 
device or in sip.conf).  I have no idea how to fix it.
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Re: [Asterisk-Users] Asterisk on Slack 10.0

[Andrew Kohlsmith]

On April 6, 2005 09:12 pm, Gary Guthary wrote:
> I just got my dev-kit & am trying to get Asterisk compiled & running on a
> Slack-10.0 box (2.4.26 kernel). - Reason is that I have 7-8 yrs experience
> on Slack & little with R/H & none with Fedora. - Besides, the docs with
> 'cutesy' little CD that came with the dev-kit don't match up with the
> Fedora release levels that are on line today. - Decided to give up & go
> back to slack-10.0 (since 2.4.26 is 'somewhat' supported).

Asterisk is very distro-agnostic, but plays best when the distro doesn't play 
silly bugger with the kernel (Gentoo, RedHat, etc.)

I have Asterisk running on Slackware 10.0 and 10.1 boxes in production.  
There's really nothing to it.  If you have any specific questions just ask.

-A.
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Re: [Asterisk-Users] Asterisk on Slack 10.0

[snacktime]

> 
> If there's anybody out there that's done something similar on a slack box,
> I'd like to talk to you.

The wiki is the first place to look for all the basic * questions,
install guides, etc..  The following url should help out.

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

Chris
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[Asterisk-Users] Receiving calls from and to H323 devices

[Guillermo Salas M.]

Hello:
This is my first message to the list.
I`ve * running with CentOS, and the h323 module loaded. I want to 
configure an Elesign 1700 that only support h323 to be able to 
communicate with other SIP devices and sip softphones. I can call from 
elesign h323 to a SIP, but can´t call from SIP to h323 device.

Greetings from Ecuador - South America.
Regards,

--
Guillermo Salas M.
Telconet S.A. Manta
Calle 15 y Av. 24 Esq.
Phone : 593 5 262 8071
Mobile: 593 9 985 5138
e-mail: [EMAIL PROTECTED]
www   : http://www.telconet.net
http://www.telcocarrier.net
Linux User: 255902
Soporte en Linea en http://www.manta.telconet.net
Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html
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[Asterisk-Users] Direct Channel Answering

[David Choo]

Dear All,

I'm trying to achieve the following with asterisk, would like to seek your
kind advise.

1: All calls to Channel 8 will go to extension 112
2: All calls to Channel 12 will go to extension 113

I've tried to read README.varibles and voip-info. It don't seem to help
much, would just like some kind soul to help me out on this. Thanks!

Here are my extension.conf entries.

[incoming]
exten = s,1,GotoIf($[${CHANNEL:8:1} = 8]?local-extensions,112,1:)
exten = s,n,GotoIf($[${CHANNEL:12:1} = 8]?local-extensions,113,1:)

Best Regards,

==
David Choo
Systems Engineer
Business & Technology Division
"Engineered for Changing Businesses"
Espore Corp Pte Ltd
68 Kallang Pudding Rd
#04-03 SYH Logistics Bldg
Singapore 349327
Tel: 65-68487806
Fax : 65-6842 2724
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[Asterisk-Users] Account Codes with SIP

[Matt Darnell]

Hello,

Does anyone know of an * plug in that will prompt a user for an
account code when they make a long distance call?

I see where you can have a static variable, but I am looking for a
lawyer bill back type application.

Thanks,
Matt
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RE: [Asterisk-Users] Asterisk on Slack 10.0

[mattf]

We have several Slack 10 boxes in production. In fact it is the recommended
distro of our astGUIclient project. We even have a full Install-from-scratch
document on our project website:

http://astguiclient.sourceforge.net/scratch_install.html

But no matter which distro you use it is always best to install Asterisk
from source.

Good luck,

MATT---

-Original Message-
From: Gary Guthary [mailto:[EMAIL PROTECTED]
Sent: Wednesday, April 06, 2005 9:13 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk on Slack 10.0


Hello -

I'm (VERY) new to the mailing list.

And after seeing some of the postings here, I can also see that I'm also
VERY new to the Asterisk game. - WOW!  I can see this'll take a while to
'catch up' (if that ever happens).

You folks have enough room for one more to play?

Without boring a lot of you & trying to practice a little -
'user-list-etiquitte', I'll keep this short.

I just got my dev-kit & am trying to get Asterisk compiled & running on a
Slack-10.0 box (2.4.26 kernel). - Reason is that I have 7-8 yrs experience
on Slack & little with R/H & none with Fedora. - Besides, the docs with
'cutesy' little CD that came with the dev-kit don't match up with the Fedora
release levels that are on line today. - Decided to give up & go back to
slack-10.0 (since 2.4.26 is 'somewhat' supported).

If there's anybody out there that's done something similar on a slack box,
I'd like to talk to you.

If I'm posting in the wrong place, I'm sorry. - Could somebody please steer
me in the right direction.

Thanks in advance.

Gary


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[Asterisk-Users] Problems with new Asterisk@Home install and Broadvoice, no incoming calls

[Craig Simon]

Hello everyone,
Let me apologize in advance.  I have spent some time googleing looking 
for my issue and as of yet been unable to resolve it.  Here is what I am 
doing.

I am installing a new AAH server for my use.  I have a Cisco 7060 and 
7920 for my eventual use.  I currently have a broadvoice account that I 
am using with a Sipura 3000 and analog phone.  I have installed AAH and 
gotten outbound calling to work perfectly. I am testing this with the 
x-lite softphone, and in can complete outgoing calls. However I can not 
get inbound calling to work at all.  Every time I try to call my number 
I immediately get the Broadvoice voicemail.

Here are my configs.
[sip.conf]
; Note: If your SIP devices are behind a NAT and your Asterisk
;  server isn't, try adding "nat=1" to each peer definition to
;  solve translation problems.
[general]
port = 5060   ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
;context = from-sip-external ; Send unknown SIP callers to this context
context=from-broadvoice
callerid = Unknown
externip=1.2.3.4
localnet=192.168.1.1/255.255.255.0
[sip_additional.conf]
register=>[EMAIL PROTECTED]:REMOVED:[EMAIL PROTECTED]/500
[500]
username=500
type=friend
secret=123
qualify=no
port=5060
pickupgroup=
nat=never
[EMAIL PROTECTED]
host=dynamic
dtmfmode=rfc2833
disallow=
context=from-internal
canreinvite=no
callgroup=
callerid="Craig Simon" <500>
allow=
[from-broadvoice]
username=9251234567
user=9251234567
type=user
secret=REMOVED
nat=yes
insecure=very
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
dtmfmode=inband
dtmf=inband
context=from-broadvoice
#type=user
[sip.broadvoice.com]
username=9251234567
type=peer
secret=REMOVED
qualify=yes
nat=yes
insecure=very
host=sip.broadvoice.com
fromuser=9251234567
fromdomain=sip.broadvoice.com
canreinvie=no
authname=9251234567
[my extensions.conf has this tidbit added to it at the top]
# my changes
[from-broadvoice]
;This extension line will ring SIP
;extension 2001 for 60 seconds then hang up. Modify as necessary to fit 
your dialplan
exten => s,1,Dial(SIP/500,60,tr)
exten => s,2,hangup

Not sure if anyone needs any additional information to figure out what I 
am screwing up here.  Any help would be greatly appreaciated.

Thanks
Craig
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RE: [Asterisk-Users] Help with simple callback application from newbie

[CM Rahman Jr.]

I am looking for same type of solution. Anybody here can help?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Herbert Chan
Sent: Wednesday, April 06, 2005 12:06 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Help with simple callback application from newbie

Hi there,

I know that this has been covered many times with already but I still
can't seem to get it to work.  Basically, all I want to do is:

1) call the asterisk server from an external line
2) Say punch in a particular extension number
3) Asterisk then hangs up and calls me back based on my Caller ID or a
pre-determined number
4) I'm then provided with a dialtone to make outgoing calls 

I wanna do this cos I'm a cheapo dude with a free-incoming call plan on
my mobile. 

I've tried and failed at using app_qcall, Its not included in the latest
CVS download.. But I chucked the files in and re-installed asterisk.  No
directory qcall exists in /var/spool/asterisk... And I don't know if I
should create one and what file should I create within that folder?

Other solutions I've searched for show scripts for CAPI call back, etc..
But I just wanna callback on a zap channel.

Help? Is there like an easy way to do this? Im a layman who never
touched linux till a week ago.. By the was, I'm using a TDM22B with 2
anolog phones attached to 2 outgoing lines

Asterisk ROCKS!!!

Herbert



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[Asterisk-Users] MWI for SER and Asterisk - ast_data vs "realtime"

[Cameron Beattie]

Bug 002980 contains a patch that delivers MWI from 
Asterisk to SIP clients registered with SER. It requires ast_data which doesn't 
patch the current CVS HEAD. Since I need MWI and I want Asterisk and SER to 
store data in a shared database I could:
- use an old version of Asterisk and install 
ast_data. The patch author runs a large-scale clustered SER/Asterisk environment 
and believes ast_data to be superior to "realtime". However "realtime" may now 
have developed to the point that these objections are no longer relevant. 
Specific problems still seem to be SIP clients behind NAT and voicemail 
notifications (both very important to me)
- use realtime and write something that gives me 
the functionality required and hope that the NAT and voicemail issues get fixed 
soon. Apart from it being a pain to write this I also have not heard of anyone 
using SER + Asterisk "realtime" in a large-scale production environment. If 
anyone has had any such successes it would be good to know about 
them.
 
I would appreciate thoughts on these (or any other) 
alternatives.
 
Regards
 
Cameron
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[Asterisk-Users] SIP Softphone for testing with Asterisk

[raymond]

Hi all,

I had just set up my asterisk server.

Can anybody know that is there any sip softphone for testing with asterisk?
(I had download some from internet but I think all are preconfig to certain
server).

Cheers.

Raymond





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Re: [Asterisk-Users] SIP Softphone for testing with Asterisk

[Brian Capouch]

raymond wrote:
Hi all,
I had just set up my asterisk server.
Can anybody know that is there any sip softphone for testing with asterisk?
(I had download some from internet but I think all are preconfig to certain
server).
That isn't a very good question.
Did you google for "asterisk soft phone?"  I got 78,000 responses, with 
virtually all of the first page being the information you are looking for.

This is a very high-volume list.  Your questions will get more friendly 
responses if it appears you have done the slightest amount of work on 
your own to see what you might find.

B.
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[Asterisk-Users] rout call from ser to asterisk

[Kamran Ahmad]

hello

i have a prblem in routing call from ser to asterisk.

i have the following senrio.

UA is registered at ser
when UA calls another UA ser try to look for the user
not found then forword the call to other side
asterisk.


problem i am facing that ser is not forwording request
to  asterisk

extensions.conf

exten=>2000,1,Dial(SIP/${EXTEN})
exten=>2000,2,Voicemail,u2000
exten=>2000,102,Voicemail,b2000

exten=>3000,1,Dial(SIP/${EXTEN})
exten=>3000,2,Voicemail,u3000
exten=>3000,102,Voicemail,b3000


ser.cfg
---
route{
lookup("location");
t_on_failure("1");
t_relay();
  
  
   if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
break;
};
if ( msg:len > max_len ) {
sl_send_reply("513", "Message too
big");
break;
};
  
  if
(loose_route()) {
t_relay();
break;
};
  
  
setflag(1);
if (method=="INVITE") record_route();
  
  
if (uri==myself) {
  
  
if (method=="REGISTER") {
  
  
   if(!radius_www_authorize("")) {
www_challenge("","0");
break;
}
  
  
save("location");
break;
};
  
  
if (!lookup("location")) {
sl_send_reply("404", "Not
Found");
break;
};
};

if (!t_relay()) {
sl_reply_error();
};
  
  
}

failure_route[1] {
  
  
if(t_check_status("486")) {
# busy, so rewrite to new location
here, then:
forward( 192.168.8.97, 5060 );
t_relay();
}
}

Thanks
Kamran



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Re: [Asterisk-Users] SIP Softphone for testing with Asterisk

[Rod Bacon]

I've tested about a dozen of them, and find firefly one of the best (others 
have more features, but I find firefly is a good mix of 
quality/features/performance). Make sure you get the third-party firefly 
though, not the one that's limited to virbiage.

Try here...
http://www.virbiage.com/firefly/download/firefly-thirdparty.exe
- Original Message - 
From: "raymond" <[EMAIL PROTECTED]>
To: 
Sent: Thursday, April 07, 2005 1:41 PM
Subject: [Asterisk-Users] SIP Softphone for testing with Asterisk


Hi all,
I had just set up my asterisk server.
Can anybody know that is there any sip softphone for testing with 
asterisk?
(I had download some from internet but I think all are preconfig to 
certain
server).

Cheers.
Raymond


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[Asterisk-Users] Any gsm -> g7231 codec translator?

[cmisip]

Is there such a thing so I can call fwd from ohphone using a low
bandwidth codec?

What program can I use to transcode the gsm sound files in asterisk 
into g7231 format?  

If there is no way to do the above, what codec do you guys recommend if
one endpoint is a dialup connection (the other broadband cable).  I am
using quicknet phonejacks on either end.




Thanks.

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Re: [Asterisk-Users] Polycom IP 600 not ringing

[Alex]

Bugger :-(
On Wed, 06 Apr 2005 20:14:06 -0500, Eric Wieling <[EMAIL PROTECTED]> wrote:
Alex wrote:
Sorry...
It should reed Polycom IP 600 does not make an  audible ring sound
 ...half a sleep :-)
 On Thu, 07 Apr 2005 10:07:57 +1000, Alex <[EMAIL PROTECTED]> wrote:
Hi guys,
Has anyone come across a problem when Polycom IP 600 does not make an   
audible ring sound, even though the call comes in? I can see it on  
LCD  and red light flashes. When I pickup the phone, everything is  
fine. It  only applies to SIP calls. If the call comes in from PSTN  
via TDM400  card, everything seems to be ok.

The same scenario happens when I try to ring from IP600 other Polycom   
phones, again they don't ring.
I am using 2.6.1 bootrom and 1.4.1 SIP firmware.

Anyone can point me in the right direction?
I have seen this problem on the Polycom 500.  ONLY happens when I call  
from port 1 of my SPA-2000, works fine on port 2.  I cannot see any  
significant difference in the way the two ports are configured (on the  
device or in sip.conf).  I have no idea how to fix it.
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