Claude- Gaelle ONGBIL
2005-May-03 00:30 UTC
[Asterisk-Users] Fwd: config for call pstn from voip
Skipped content of type multipart/alternative-------------- next part -------------- An embedded message was scrubbed... From: Claude- Gaelle ONGBIL <ongcla@yahoo.fr> Subject: config for call pstn from voip Date: Mon, 2 May 2005 12:03:05 +0200 (CEST) Size: 3785 Url: http://lists.digium.com/pipermail/asterisk-users/attachments/20050503/a20d8526/attachment.eml
Hi Claude. I just have giving some advices to someone with your same
problem. I assume you have the analog phone you want to call, behind
some AnalogPBX, then you have to call the analogPBX and tell him that
you want to call some analog extension. How?
Well, im just going to paste the same response i put a couple of
minuts ago, so you dont have to search it
//////////// RESPONSE TO julio ////////////////////////////////
Hi Julio. It would be nice if you show the extensions.conf that
handles that kind of calls. You can do something like this:
[macro-analogpbx]
exten => s,1,Cut(ChannelType=CHANNEL,/,1) //check if the call comes
from other Zap ch
exten => s,2,GotoIf($[${ChannelType} = Zap] ? 3 : 6) //If does, go 3,
othewise 6
exten => s,3,Flash()
exten => s,4,SendDTMF(${analogprefix}${num}) //send the DTMF for the
extension dialed
exten => s,5,Hangup()
exten => s,6,Dial(Zap/g${analoggroup}/${analogprefix}${num}) //if the
call comes from SIP or IAX then execute Dial trough some group in
zapata
exten => s,7,Hangup()
You can see some variables i just use for administration of my PBX,
but i hope you understand the concept.
Good Look
////////////////////// END RESPONSE TO julio
Ok, so, hope it helps you too. I does not, try being more explicit
about the problem.
Best Regards.
-moy
On 5/3/05, Claude- Gaelle ONGBIL <ongcla@yahoo.fr>
wrote:>
>
> Note: forwarded message attached.
>
> ________________________________
> D?couvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour
vos
> mails !
> Cr?ez votre Yahoo! Mail
>
>
>
> ---------- Forwarded message ----------
> From: Claude- Gaelle ONGBIL <ongcla@yahoo.fr>
> To: asterisk-users@lists.digium.com
> Date: Mon, 2 May 2005 12:03:05 +0200 (CEST)
> Subject: config for call pstn from voip
>
>
>
>
>
> hello,
>
>
>
> newasterisk user i've configured my 2 sip phones and they can place
calls
> .i,ve also fxo card and i've configured channel ;now it's possible
to
> recieve analog calls with my sip phone but i want to make call with my sip
> phone to analog it's possible?
> when i dial a number my sip phone answer the call and i've echo please
may
> somebody help me?
>
>
> there is my config file
>
> zaptel.conf
> fxsls=4#X100P
> defaultzone=fr
> loadzone=fr
>
> zapata.conf
> [channels]
> language=fr
> relaxdtmf=yes
> immediate=no
> context=pstn
> signalling=fxs_ls;X100P
> ;Cidsignalling=v23
> ;Cidstart=polarity
> ;usecallerid=yes
> ;callerid="fone" <60
>
> extensions.conf
> [general]
> static=yes
> writeprotect=no
> [pstn]
> exten => 19100,1,dial(SIP/799&SIP/788)
> exten => 788,1,dial(SIP/788:5060)
> exten => 799,1,dial(SIP/799:5060)
> exten => _00NXXXXXXXX,1,dial(Zap/4/${EXTEN:1}); i want to
> call analog phone
> exten => _6059,1,dial(SIP/799)
> exten => s,1,dial(SIP/799&SIP/788);here i can recieve analog calls
>
>
>
>
>
>
> regards.
>
>
> ________________________________
>
>
>
>
>
>
> ________________________________
> D?couvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour
vos
> mails !
> Cr?ez votre Yahoo! Mail
>
>
> _______________________________________________
> 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
>
>
--
"Su nombre es GNU/Linux, no solamente Linux, mas info en
http://www.gnu.org"