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: asterisk-users@lists.digium.com 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
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.
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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: asterisk-users@lists.digium.com 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] 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 sip:[EMAIL PROTECTED]:5060
-- 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 sip:[EMAIL PROTECTED]:5060 -- 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 ___ 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
