asterisk users - Oct 2008 - Panasonic x Asterisk if I can emulate Panasonic fast!

Im a 3-days-asterisk-newbie. In 3 weeks, I must have a PBX installed in
a new office of ours: Panasonic or Asterisk. Asterisk would be, if I can
emulate some Panasonic functions on Asterisk fast, to convince the
executives.

What I have done until now: Bought 1 Linksys pap2 (2 FXS), 1 Linksys
SPA3102 (1 FXS + 1 FXO) for making asterisk tests. Configured
Asterisk/Fedora 9 so I can make SIP->PSTN and PSTN->SIP calls.

Works. Now, I need this help, please:

* Dialing from inside (pap2-FXS connected phone) to another number on
the same city (goes out by SPA3102 FXO), voice works fine. But when a
menu answers, and I dial over, the menu dialed keys works only 20% of
all times. Why could this would be? Voltage levels? sound gains? Dialed
keys get distorsioned when passing over the 2 Linksys? Linksys or
Asterisk swallowing some dialed key? I noticed some echo...

* I need to assign two codes to each user, one for international calls
charged to the office, another for international calls charged to the
user. If the user enters an incorrect code, the call should not proceed.

* I need to get a formatted calls report for the administrators to
charge the users.

I just am confused and stucked with all the documentation in Internet,
and all this new asterisk jargon. I just need some links (or some
directions) to go fast on this topics. Of course, some more help would
be appreciated.

Thanks a lot.
-- 
Rodolfo Alcazar
Responsable red y datos

Deutsche Gesellschaft f?r
Technische Zusammenarbeit (GTZ) GmbH

Programa de Apoyo a la Gesti?n P?blica Descentralizada y
Lucha Contra La Pobreza - PADEP
Av. S?nchez Lima 2226
La Paz, Bolivia

Tel: +591 22417628 (121)
Fax: +591 22417628 (126)
Web: www.padep.org.bo
Email: rodolfo.alcazar at padep.org.bo

Rodolfo Alcazar Portillo wrote:
> Im a 3-days-asterisk-newbie. In 3 weeks, I must have a PBX installed in
> a new office of ours: Panasonic or Asterisk. Asterisk would be, if I can
> emulate some Panasonic functions on Asterisk fast, to convince the
> executives.
>   
Asterisk is more featured than Panasonic, but you must to know Asterisk
to convince your executives.... ;-)
> What I have done until now: Bought 1 Linksys pap2 (2 FXS), 1 Linksys
> SPA3102 (1 FXS + 1 FXO) for making asterisk tests. Configured
> Asterisk/Fedora 9 so I can make SIP->PSTN and PSTN->SIP calls.
>
> Works. Now, I need this help, please:
>
> * Dialing from inside (pap2-FXS connected phone) to another number on
> the same city (goes out by SPA3102 FXO), voice works fine. But when a
> menu answers, and I dial over, the menu dialed keys works only 20% of
> all times. Why could this would be? Voltage levels? sound gains? Dialed
> keys get distorsioned when passing over the 2 Linksys? Linksys or
> Asterisk swallowing some dialed key? I noticed some echo...
>   
Probably you are sending dtmf signals inband. Try outband.
For the echo, try to change the FXO/FXS impedance, and/or playing with
the rx and tx gains. I assume that do you have echo cancelling enable in
both SPA.
> * I need to assign two codes to each user, one for international calls
> charged to the office, another for international calls charged to the
> user. If the user enters an incorrect code, the call should not proceed.
>   
See account codes. You can start here:
http://www.voip-info.org/wiki-Asterisk+Billing
> * I need to get a formatted calls report for the administrators to
> charge the users.
>   
See same link, or google for billing
> I just am confused and stucked with all the documentation in Internet,
> and all this new asterisk jargon. I just need some links (or some
> directions) to go fast on this topics. Of course, some more help would
> be appreciated.
>   
The link to start:
http://www.voip-info.org
> Thanks a lot.
>   
De nada

Jorge

Being a Panasonic dealer and having more than 50 Asterisk system in
production, I can tell you that if this is your first Asterisk
project, then go with Panasonic, you'll safe yourself lots of
aggravation and have a happier customer.

Some features of the Panasonic you will never be able to emulate on Asterisk.

While depending on the needs of that customer, and in some cases I
would suggest dive into Asterisk, I gather from the subject (yes I
have read the whole message, for those of you out there that might
think that I did not) that a Panasonic will work nicely for them,
therefore my advice stick with Panasonic.

On Mon, Oct 13, 2008 at 9:15 PM, Rodolfo Alcazar Portillo
<rodolfo.alcazar at padep.org.bo> wrote:
> Im a 3-days-asterisk-newbie. In 3 weeks, I must have a PBX installed in
> a new office of ours: Panasonic or Asterisk. Asterisk would be, if I can
> emulate some Panasonic functions on Asterisk fast, to convince the
> executives.
>
> What I have done until now: Bought 1 Linksys pap2 (2 FXS), 1 Linksys
> SPA3102 (1 FXS + 1 FXO) for making asterisk tests. Configured
> Asterisk/Fedora 9 so I can make SIP->PSTN and PSTN->SIP calls.
>
> Works. Now, I need this help, please:
>
> * Dialing from inside (pap2-FXS connected phone) to another number on
> the same city (goes out by SPA3102 FXO), voice works fine. But when a
> menu answers, and I dial over, the menu dialed keys works only 20% of
> all times. Why could this would be? Voltage levels? sound gains? Dialed
> keys get distorsioned when passing over the 2 Linksys? Linksys or
> Asterisk swallowing some dialed key? I noticed some echo...
>
> * I need to assign two codes to each user, one for international calls
> charged to the office, another for international calls charged to the
> user. If the user enters an incorrect code, the call should not proceed.
>
> * I need to get a formatted calls report for the administrators to
> charge the users.
>
> I just am confused and stucked with all the documentation in Internet,
> and all this new asterisk jargon. I just need some links (or some
> directions) to go fast on this topics. Of course, some more help would
> be appreciated.
>
> Thanks a lot.
> --
> Rodolfo Alcazar
> Responsable red y datos
>
> Deutsche Gesellschaft f?r
> Technische Zusammenarbeit (GTZ) GmbH
>
> Programa de Apoyo a la Gesti?n P?blica Descentralizada y
> Lucha Contra La Pobreza - PADEP
> Av. S?nchez Lima 2226
> La Paz, Bolivia
>
> Tel: +591 22417628 (121)
> Fax: +591 22417628 (126)
> Web: www.padep.org.bo
> Email: rodolfo.alcazar at padep.org.bo
>
>
> _______________________________________________
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

Am Freitag, den 17.10.2008, 13:22 +0800 schrieb Cindy
Tan:
> HI.... this is cindy... i am still a student... i want to learn more
> things about asterisk from you. can i ask you something? 
Yes. CC to the list, expecting qualified answers :)
> actually, i am thinking how live messager can works on asterisk. 
As I said, I'm also an asterisk newbie, but still can help. What you
should check is if livemsn supports some standard protocol, as jabber,
sip, h323 (I don't use windows, don't know what liveMSN can do). Or if
both have some protocol in common. Then, configure accounts on asterisk,
and connect liveMSNs to asterisk. I don't believe liveMSN is that open.
Skype is releasing a module for asterisk. Maybe MS will follow.
 
> I want things to works on calling to and from messager and soft
> phone. 
"Asterisk can bridge and translate different types of VoIP protocols
like SIP, MGCP, and H.323", says some review. Well, you must just try to
connect MSN with asterisk. Probably asterisk handles communication
between devices transparently. 

Tell us if you make it. Good luck.
-- 
Rodolfo Alcazar
Responsable red y datos

Deutsche Gesellschaft f?r
Technische Zusammenarbeit (GTZ) GmbH

Programa de Apoyo a la Gesti?n P?blica Descentralizada y
Lucha Contra La Pobreza - PADEP
Av. S?nchez Lima 2226
La Paz, Bolivia

Tel: +591 22417628 (121)
Fax: +591 22417628 (126)
Web: www.padep.org.bo
Email: rodolfo.alcazar at padep.org.bo

Can i install Asterisk beside Nortel PCM, just for recording all calls on E1
(incoming and outgoing calls)
I want to get the E1 into Asterisk (Digium)

how can this scenario be achieved in details please ?


> Date: Sat, 25 Oct 2008 07:42:09 +1300
> From: r.scobie at clear.net.nz
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] Panasonic x Asterisk ... NO PROBLEM!
> 
> 
> 
> Jonn R Taylor wrote:
> > Install a T1 between the Panasonic and Asterisk and program the T1 in
the Panasonic as a other custom PBX. VOIP card would be the best.
> > 
> > Jonn
> 
> One thing to beware of with the Panasonic VoIP card, is that I have 
> found no way of getting it to pass out of band DTMF, possibly because it 
> handles this in a proprietary way.
> 
> This has been my experience with a TDA100 and VoIP card.
> 
> Regards,
> 
> Richard
> 
> _______________________________________________
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> 
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You could, under programming section 1.3.4 in the http interface to
configure the GW card enable DTMF Detection, that will enable Out of
Band DTMF. In the TDE they renamed this to DTMF signalling.

On Fri, Oct 24, 2008 at 2:42 PM, Richard Scobie <r.scobie at clear.net.nz>
wrote:
>
>
> Jonn R Taylor wrote:
>> Install a T1 between the Panasonic and Asterisk and program the T1 in
the Panasonic as a other custom PBX. VOIP card would be the best.
>>
>> Jonn
>
> One thing to beware of with the Panasonic VoIP card, is that I have
> found no way of getting it to pass out of band DTMF, possibly because it
> handles this in a proprietary way.
>
> This has been my experience with a TDA100 and VoIP card.
>
> Regards,
>
> Richard
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

C F wrote:
> You could, under programming section 1.3.4 in the http interface to
> configure the GW card enable DTMF Detection, that will enable Out of
> Band DTMF. In the TDE they renamed this to DTMF signalling.
Believe me, I spent a great deal of time on this including Ethereal 
captures and nothing worked.

Have you succeeded with this?

If so, what DTMF protocols were passed?

Regards,

Richard

I have never had to play around with this, as I only used that card to
interconnect 2 Panasonic systems which is proprietary although based
on h323, and never for a provider based system.

On Mon, Oct 27, 2008 at 12:56 AM, Richard Scobie <r.scobie at
clear.net.nz> wrote:
>
>
> C F wrote:
>> You could, under programming section 1.3.4 in the http interface to
>> configure the GW card enable DTMF Detection, that will enable Out of
>> Band DTMF. In the TDE they renamed this to DTMF signalling.
>
> Believe me, I spent a great deal of time on this including Ethereal
> captures and nothing worked.
>
> Have you succeeded with this?
>
> If so, what DTMF protocols were passed?
>
> Regards,
>
> Richard
>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>