Re: [Asterisk-Users] asterisk to analog pbx
Well , problem solved the problem was with [EMAIL PROTECTED] i have installed an asterisk from scratch and everything works fine now .. weird ./ Thanks! El mié, 04-05-2005 a las 10:23 +0200, Julio Saura escribió: > Hi > i posted it this morning > > i guess is a [EMAIL PROTECTED] problem... installing a new OS with * from > scratch > > it does not even call outside connecting fxo to pots :? > > > > > > El mié, 04-05-2005 a las 09:55 +0200, Mehdi Chouikh escribió: > > Hello > > all is right, the analog extension should ring, but maybe your dialplan is > > not correct or you call a bad extension in you PBX. > > can you post your dialplan?, to see it. > > regards > > - Original Message - > > From: "Julio Saura" <[EMAIL PROTECTED]> > > To: > > Sent: Tuesday, May 03, 2005 2:37 PM > > Subject: [Asterisk-Users] asterisk to analog pbx > > > > > > > Hi there > > > > > > i have an asterisk box running ok, and now i am trying to integrate it > > > with my local analog pbx > > > > > > So far, i have connected the fxo port of my * to an analog extension > > > port of my analog pbx. > > > > > > As far as i know, if a call an extension of my analog pbx on a sip phone > > > ( i have done the right dial plan for routing these calls to de zap > > > channel ) the analog pbx extension should ring ... > > > > > > am i right? > > > > > > asterisk says the call is done, but the analog extension keeps in > > > silence .. :? > > > > > > any clue, am i doing something wrong? > > > > > > Best regards. > > > > > > ___ > > > 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 > > > > ___ > > 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 > > ___ > 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 ___ 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
Re: [Asterisk-Users] asterisk to analog pbx
Hi i posted it this morning i guess is a [EMAIL PROTECTED] problem... installing a new OS with * from scratch it does not even call outside connecting fxo to pots :? El mié, 04-05-2005 a las 09:55 +0200, Mehdi Chouikh escribió: > Hello > all is right, the analog extension should ring, but maybe your dialplan is > not correct or you call a bad extension in you PBX. > can you post your dialplan?, to see it. > regards > - Original Message ----- > From: "Julio Saura" <[EMAIL PROTECTED]> > To: > Sent: Tuesday, May 03, 2005 2:37 PM > Subject: [Asterisk-Users] asterisk to analog pbx > > > > Hi there > > > > i have an asterisk box running ok, and now i am trying to integrate it > > with my local analog pbx > > > > So far, i have connected the fxo port of my * to an analog extension > > port of my analog pbx. > > > > As far as i know, if a call an extension of my analog pbx on a sip phone > > ( i have done the right dial plan for routing these calls to de zap > > channel ) the analog pbx extension should ring ... > > > > am i right? > > > > asterisk says the call is done, but the analog extension keeps in > > silence .. :? > > > > any clue, am i doing something wrong? > > > > Best regards. > > > > ___ > > 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 > > ___ > 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 ___ 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
Re: [Asterisk-Users] asterisk to analog pbx
Hi
this is the macro used for that purpose ..
[macro-dialout-trunk]
exten => s,1,GotoIf($[foo${ECID${CALLERIDNUM}} = foo]?4);check
for CID override for exten
exten => s,2,SetCallerID(${ECID${CALLERIDNUM}})
exten => s,3,Goto(6)
exten => s,4,GotoIf($[foo${OUTCID_${ARG1}} = foo]?6);check
for CID override for trunk
exten => s,5,SetCallerID(${OUTCID_${ARG1}})
exten => s,6,SetGroup(OUT_${ARG1})
exten => s,7,CheckGroup(${OUTMAXCHANS_${ARG1}})
; if we've used up the max channels, continue at 108 (n+101)
exten => s,8,SetVar(DIAL_NUMBER=${ARG2})
exten => s,9,SetVar(DIAL_TRUNK=${ARG1})
exten => s,10,AGI(fixlocalprefix) ; this sets DIAL_NUMBER to the proper
dial string for this trunk
exten => s,11,Dial(${OUT_${ARG1}}/${OUTPREFIX_${ARG1}}${DIAL_NUMBER})
; if dial fails (ie, all channels are busy), continue at 112 (n+101)
;exten => s,11,Dial(Zap/0/${DIAL_NUMBER})
; we should only get here if the call was successful (?)
exten => s,9,Congestion
; exit points for macro
exten => s,108,NoOp(max channels used up)
exten => s,112,NoOp(dial failed)
as u can see is also a dial instruction
the call seems to be done but in fact my analog extension does not
ring :/
any clue?
Thanks again
El mar, 03-05-2005 a las 10:17 -0500, Moises Silva escribió:
> Hi Julio. It would be nice if you show the extensions.conf that
> handles that kind of calls. You can do something like this:
>
> [macro-analogpbx]
> exten => s,1,Cut(ChannelType=CHANNEL,/,1) //check if the call comes
> from other Zap ch
> exten => s,2,GotoIf($[${ChannelType} = Zap] ? 3 : 6) //If does, go 3,
> othewise 6
> exten => s,3,Flash()
> exten => s,4,SendDTMF(${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
>
> - moy
>
> On 5/3/05, Julio Saura <[EMAIL PROTECTED]> wrote:
> > Hi there
> >
> > i have an asterisk box running ok, and now i am trying to integrate it
> > with my local analog pbx
> >
> > So far, i have connected the fxo port of my * to an analog extension
> > port of my analog pbx.
> >
> > As far as i know, if a call an extension of my analog pbx on a sip phone
> > ( i have done the right dial plan for routing these calls to de zap
> > channel ) the analog pbx extension should ring ...
> >
> > am i right?
> >
> > asterisk says the call is done, but the analog extension keeps in
> > silence .. :?
> >
> > any clue, am i doing something wrong?
> >
> > Best regards.
> >
> > ___
> > 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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[Asterisk-Users] asterisk to analog pbx
Hi there i have an asterisk box running ok, and now i am trying to integrate it with my local analog pbx So far, i have connected the fxo port of my * to an analog extension port of my analog pbx. As far as i know, if a call an extension of my analog pbx on a sip phone ( i have done the right dial plan for routing these calls to de zap channel ) the analog pbx extension should ring ... am i right? asterisk says the call is done, but the analog extension keeps in silence .. :? any clue, am i doing something wrong? Best regards. ___ 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
Re: [Asterisk-Users] Asterisk PBX with X100P in India
what kind of problems do u have? can u explain more in detail so we can try helping you? best regards El vie, 15-04-2005 a las 01:29 -0700, Min Hwan Chang escribió: > I'm currently trying to set up an Asterisk PBX system in India. > However I'm having trouble configuring the X100P to dial out on the > POTS line. Does anyone have any knowledge about this? > > I know the telephone system is a bit different in India, so would the > X100P not be suitable? Is there a change I need to make in the > Zaptel.conf or zapata.conf? > > Because I'm using [EMAIL PROTECTED] 0.6, the extensions.conf is also > pretty frustrating... > > Any help here would be appreciated. > ___ > 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 ___ 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
Re: [Asterisk-Users] Problem with fxo
Hi Moises
thanks for the help
but i have the same problem
exten => _9X.,1,Dial(Zap/g1/${EXTEN:1})
this is my extension for dialing out
still the same weird exception 15 error :/
and group 1 es the one on my zapata.conf
starting to think about hardware problem :/
El mar, 12-04-2005 a las 14:20 +, Moises Silva escribió:
> I have no Idea of the strange errors, but as far as i know, the proper
> way of calling is:
>
> Zap/g${group}/${phone_number}
>
> where ${group} is a valid group inside zapata.conf, and
> ${phone_number} is the desired PSTN phone to call. In you email you
> wrote the messages and i can see that you missed the letter 'g'
> before the group and the last '/' slash. Give that a try, may be will
> work.
>
> Best Regards
>
> - Moy
>
> On Apr 12, 2005 11:23 AM, Julio Saura <[EMAIL PROTECTED]> wrote:
> > Hi,
> >
> > i am trying to use my fxo card for analog calls ..
> >
> > fxo card seems to be ok, working properly but when trying to call
> > outside ( from a sip phone ot pstn ) i get the following error on
> > asterisk .
> >
> > Apr 12 11:59:24 DEBUG[4231]: chan_sip.c:4633 build_route: build_route:
> > Contact hop: Drugo
> > -- Executing Dial("SIP/69-562c", "Zap/1/651559526|5") in new stack
> > Apr 12 11:59:24 DEBUG[4231]: chan_zap.c:1645 zt_call: Dialing
> > '651559526'
> > Apr 12 11:59:24 DEBUG[4231]: chan_zap.c:1706 zt_call: Deferring
> > dialing...
> > -- Called 1/651559526
> > Apr 12 11:59:25 DEBUG[4231]: chan_zap.c:3770 __zt_exception: Exception
> > on 15, channel 1
> > Apr 12 11:59:25 DEBUG[4231]: chan_zap.c:3082 zt_handle_event: Got event
> > Hook Transition Complete(12) on channel 1 (index 0)
> > Apr 12 11:59:27 DEBUG[4231]: chan_zap.c:3770 __zt_exception: Exception
> > on 15, channel 1
> > Apr 12 11:59:27 DEBUG[4231]: chan_zap.c:3082 zt_handle_event: Got event
> > Dial Complete(9) on channel 1 (index 0)
> > Apr 12 11:59:27 DEBUG[4231]: chan_zap.c:1224 zt_enable_ec: Enabled echo
> > cancellation on channel 1
> > Apr 12 11:59:27 DEBUG[4231]: channel.c:1363 ast_read: Dropping duplicate
> > answer!
> >
> > any clue?
> >
> > got no info about exception 15 :/
> >
> > Thanks in advance
> >
> > ___
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[Asterisk-Users] Problem with fxo
Hi,
i am trying to use my fxo card for analog calls ..
fxo card seems to be ok, working properly but when trying to call
outside ( from a sip phone ot pstn ) i get the following error on
asterisk .
Apr 12 11:59:24 DEBUG[4231]: chan_sip.c:4633 build_route: build_route:
Contact hop: Drugo
-- Executing Dial("SIP/69-562c", "Zap/1/651559526|5") in new stack
Apr 12 11:59:24 DEBUG[4231]: chan_zap.c:1645 zt_call: Dialing
'651559526'
Apr 12 11:59:24 DEBUG[4231]: chan_zap.c:1706 zt_call: Deferring
dialing...
-- Called 1/651559526
Apr 12 11:59:25 DEBUG[4231]: chan_zap.c:3770 __zt_exception: Exception
on 15, channel 1
Apr 12 11:59:25 DEBUG[4231]: chan_zap.c:3082 zt_handle_event: Got event
Hook Transition Complete(12) on channel 1 (index 0)
Apr 12 11:59:27 DEBUG[4231]: chan_zap.c:3770 __zt_exception: Exception
on 15, channel 1
Apr 12 11:59:27 DEBUG[4231]: chan_zap.c:3082 zt_handle_event: Got event
Dial Complete(9) on channel 1 (index 0)
Apr 12 11:59:27 DEBUG[4231]: chan_zap.c:1224 zt_enable_ec: Enabled echo
cancellation on channel 1
Apr 12 11:59:27 DEBUG[4231]: channel.c:1363 ast_read: Dropping duplicate
answer!
any clue?
got no info about exception 15 :/
Thanks in advance
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