AW: [Asterisk-Users] Dial out on Zap: Can't fix up channel from 31 to 30 because 30 is already in use??
I'm not sure if it's the same problem but your error message likely the same.
after i additing pridialplan=local in the zapata.conf i'm able to make
outboundcalls (located in germany)
marcus
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Gesendet: Dienstag, 11. April 2006 16:33
An: asterisk-users@lists.digium.com
Betreff: [Asterisk-Users] Dial out on Zap: Can't fix up channel from 31 to 30
because 30 is already in use??
Hi,
I still cant dial out on Zap and I really have no clue why.
I'm using Asterisk Asterisk 1.2.6 and Zaptel 1.2.5 with Digium card 4 ports, 31
channels each and able to receive incoming calls and fax perfectly.
I've done this in my dial plan.
exten = 111,1,Answer()
exten = 111,n,Ringing()
exten = 111,n,Wait(2)
exten = 111,n,AbsoluteTimeout(30)
exten = 111,n,Dial(Zap/G1/002212601574) exten = 111,n,NoOp(${DIALSTATUS})
exten = 111,n,Busy() exten = 111,n,Hangup()
My zapata.conf is like this
[channels]
context=from-pstn
group=0
switchtype=euroisdn
overlapdial=yes
faxdetect=no
echocancel=yes
echocancelwhenbridged=yes
; PRI port 1 (E1)
; context=1
group=1
signalling=pri_cpe
channel=1-15,17-31
And I've got this on my CLI:
-- Accepting overlap call from '2212601571' to '111' on channel 0/31, span 1
-- Starting simple switch on 'Zap/31-1'
-- Executing Answer(Zap/31-1, ) in new stack
-- Executing Ringing(Zap/31-1, ) in new stack
-- Executing Wait(Zap/31-1, 2) in new stack
-- Executing AbsoluteTimeout(Zap/31-1, 30) in new stack
-- Set Absolute Timeout to 30
-- Executing Dial(Zap/31-1, Zap/G1/002212601574) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G1/002212601574
-- Moving call from channel 31 to channel 30 Apr 11 16:27:06
WARNING[10322]: chan_zap.c:7745 pri_fixup_principle:
Can't fix up channel from 31 to 30 because 30 is already in use Apr 11 16:27:06
WARNING[10322]: chan_zap.c:9046 pri_dchannel: Unable to move channel 30!
-- Channel 0/30, span 1 got hangup request Apr 11 16:27:06 WARNING[10966]:
app_dial.c:706 wait_for_answer: Unable to forward voice
-- Hungup 'Zap/30-1'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing NoOp(Zap/31-1, CHANUNAVAIL) in new stack
-- Executing Busy(Zap/31-1, ) in new stack
-- Channel 0/31, span 1 got hangup request
== Spawn extension (from-pstn, 111, 7) exited non-zero on 'Zap/31-1'
-- Executing NoOp(Zap/31-1, ) in new stack
-- Executing Goto(Zap/31-1, 999) in new stack
-- Goto (from-pstn,h,999)
-- Hungup 'Zap/31-1'
Could somebody give me a clue?
Pim
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Re: AW: [Asterisk-Users] Dial out on Zap: Can't fix up channel from 31 to 30 because 30 is already in use??
Hi Marcus
Yesterday I tried that but it didn't work but today I tried again just
as u said and it works!!
Danke schön! Vielen Dank!
Gruß,
Pim
[EMAIL PROTECTED] wrote:
I'm not sure if it's the same problem but your error message likely the same.
after i additing pridialplan=local in the zapata.conf i'm able to make outboundcalls (located in germany)
marcus
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Gesendet: Dienstag, 11. April 2006 16:33
An: asterisk-users@lists.digium.com
Betreff: [Asterisk-Users] Dial out on Zap: Can't fix up channel from 31 to 30
because 30 is already in use??
Hi,
I still cant dial out on Zap and I really have no clue why.
I'm using Asterisk Asterisk 1.2.6 and Zaptel 1.2.5 with Digium card 4 ports, 31
channels each and able to receive incoming calls and fax perfectly.
I've done this in my dial plan.
exten = 111,1,Answer()
exten = 111,n,Ringing()
exten = 111,n,Wait(2)
exten = 111,n,AbsoluteTimeout(30)
exten = 111,n,Dial(Zap/G1/002212601574) exten = 111,n,NoOp(${DIALSTATUS}) exten
= 111,n,Busy() exten = 111,n,Hangup()
My zapata.conf is like this
[channels]
context=from-pstn
group=0
switchtype=euroisdn
overlapdial=yes
faxdetect=no
echocancel=yes
echocancelwhenbridged=yes
; PRI port 1 (E1)
; context=1
group=1
signalling=pri_cpe
channel=1-15,17-31
And I've got this on my CLI:
-- Accepting overlap call from '2212601571' to '111' on channel 0/31, span 1
-- Starting simple switch on 'Zap/31-1'
-- Executing Answer(Zap/31-1, ) in new stack
-- Executing Ringing(Zap/31-1, ) in new stack
-- Executing Wait(Zap/31-1, 2) in new stack
-- Executing AbsoluteTimeout(Zap/31-1, 30) in new stack
-- Set Absolute Timeout to 30
-- Executing Dial(Zap/31-1, Zap/G1/002212601574) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G1/002212601574
-- Moving call from channel 31 to channel 30 Apr 11 16:27:06 WARNING[10322]: chan_zap.c:7745 pri_fixup_principle:
Can't fix up channel from 31 to 30 because 30 is already in use Apr 11 16:27:06 WARNING[10322]: chan_zap.c:9046 pri_dchannel: Unable to move channel 30!
-- Channel 0/30, span 1 got hangup request Apr 11 16:27:06 WARNING[10966]:
app_dial.c:706 wait_for_answer: Unable to forward voice
-- Hungup 'Zap/30-1'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing NoOp(Zap/31-1, CHANUNAVAIL) in new stack
-- Executing Busy(Zap/31-1, ) in new stack
-- Channel 0/31, span 1 got hangup request
== Spawn extension (from-pstn, 111, 7) exited non-zero on 'Zap/31-1'
-- Executing NoOp(Zap/31-1, ) in new stack
-- Executing Goto(Zap/31-1, 999) in new stack
-- Goto (from-pstn,h,999)
-- Hungup 'Zap/31-1'
Could somebody give me a clue?
Pim
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AW: [Asterisk-Users] Dial out on Zap
Hi, i was able to fix this problem when i added the line pridialplan=local in the zapata.conf but it depends on your telco, i think. marcus -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 6. April 2006 11:50 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] Dial out on Zap Hi, I'm trying to test my dial out function so I did something like this in extensions.conf exten = 999,1,Dial(Zap/g1/02601591) exten = 999,102,Congestion() My Zapata.conf looks something like this [channels] context=from-pstn group=0 switchtype=euroisdn overlapdial=yes faxdetect=no ; PRI port 1 (E1) ; context=1 group=1 signalling=pri_cpe channel=1-15,17-31 I am able to receive the fax just fine with this setting. So I think it's ok. I'm using a Digium card connecting to a PSTN. There're 4 ports on the card, 31 channels each, but we currently use one. When I call extension 999, it was supposed to forward my call to 02601591, right? But it didn't. It just gets silence and it hangs up the call. On my CLI it looks something like this: -- Executing Dial(Zap/1-1, Zap/g1/02601591) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/02601591 -- Moving call from channel 1 to channel 2 Apr 6 11:08:08 WARNING[5854]: chan_zap.c:7745 pri_fixup_principle: Can't fix up channel from 1 to 2 because 2 is already in use Apr 6 11:08:08 WARNING[5854]: chan_zap.c:9046 pri_dchannel: Unable to move channel 2! -- Zap/2-1 is proceeding passing it to Zap/1-1 Apr 6 11:08:22 NOTICE[5849]: chan_iax2.c:5691 update_registry: Restricting registration for peer 'hylafax-iaxmodem' to 60 seconds (requested 300) -- Channel 0/2, span 1 got hangup request - I didn't hang up the call. It did by itself. -- Hungup 'Zap/2-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing Congestion(Zap/1-1, ) in new stack -- Channel 0/1, span 1 got hangup request == Spawn extension (voice, 999, 102) exited non-zero on 'Zap/1-1' -- Executing SetVar(Zap/1-1, HANGUP_TIME=1144314509) in new stack -- Executing NoOp(Zap/1-1, 16) in new stack -- Hungup 'Zap/1-1' I'm a bit confused what I did wrong. Do I need a second line or something?? Pim ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
