[asterisk-users] outgoing call problem
Hi. I'm having a bit of trouble with outgoing calls on zap channels. When i try to make an outgoing call asterisk doesn't detect if the other party answers. When i run 'show channels verbose' in CLI asterisk tells me that the respective channles are in ringing state like this: Channel Context Extension Prio State Application Data CallerID Duration Accountcode BridgedTo Zap/19-1 agentie s 1 Dialing AppDial (Outgoing Line) 00726710704 (None) Zap/15-1 int_omg 00726710704 5 Ring Dial Zap/g5/0726710704||T 00:00:14 (None) although i can speak to the called party. ___ --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
Re: [asterisk-users] outgoing call problem
Zap channels consider the call answered when dialing is complete, at least with the analog interface. There is no answer supervision provided to the PSTN with a POTS line Don't know if this extends to a PRI or not. John Novack Alexandru Voinescu wrote: Hi. I'm having a bit of trouble with outgoing calls on zap channels. When i try to make an outgoing call asterisk doesn't detect if the other party answers. When i run 'show channels verbose' in CLI asterisk tells me that the respective channles are in ringing state like this: Channel Context Extension Prio State Application Data CallerID Duration Accountcode BridgedTo Zap/19-1 agentie s 1 Dialing AppDial (Outgoing Line) 00726710704 (None) Zap/15-1 int_omg 00726710704 5 Ring Dial Zap/g5/0726710704||T 00:00:14 (None) although i can speak to the called party. ___ --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
Re: [asterisk-users] outgoing call problem
Hi Marco,
Thanks for your reply. Dial peer is working normal, but i heard horrible
noise instead ring tone.
Is my digium tdm04b card has problem? I have tested tdm04b using zttest. It
seems is working normal. Would you give me advice?
Thanks for help
Here is some test using zttest
zttest
Opened pseudo zap interface, measuring accuracy...
99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%
99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%
--- Results after 14 passes ---
Best: 99.987793 -- Worst: 99.987793 -- Average: 99.987793
Heere is some debug message.
-- Registered SIP '1000' at 192.168.0.12 port 4910 expires 3600
-- Saved useragent X-Lite release 1002tx stamp 29712 for peer 1000
-- Executing Answer(SIP/1000-081ac318, ) in new stack
-- Executing Dial(SIP/1000-081ac318, Zap/g1/23) in new stack
-- Called g1/23
-- Zap/1-1 answered SIP/1000-081ac318
-- Hungup 'Zap/1-1'
== Spawn extension (home, 923, 2) exited non-zero on 'SIP/1000-081ac318'
Regards,
Ganbaa
- Original Message -
From: Marco Mouta [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, July 10, 2006 10:31 PM
Subject: Re: [asterisk-users] outgoing call problem
I'm not a a guru, but
Check this line:
exten = _9.,2,Dial(Zap/g1/${EXTEN})
do you really want to dial digit 9 through your ZapLine? are you
connected to another pbx?
If you don't want do dial 9 to PSTN line , but you want your users to
dial 9 to place outgoing calls, try this:
exten = _9.,2,Dial(Zap/g1/${EXTEN:1})
Hope it helps.
Ps. Give me some feedback if you solved the problem
On 7/10/06, Ganbaa [EMAIL PROTECTED] wrote:
Hi,
I have configured digium tdm04b card with asterisk on debian. Incoming
call
is ok. But outgoing call has problem. Would you give me advice ?
Here is my config files.
zaptel.conf
fxsks=1
fxsks=2
fxsks=3
fxsks=4
loadzone=us
defaultzone=us
zapata.conf
[channels]
language=en
context=incoming
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
transfer=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=1
txgain=4
group=1
callgroup=1
pickupgroup=1
immediate=no
musiconhold=default
busydetect=yes
callprogress=no
channel = 1-4
extension.conf
[general]
static=yes
writeprotect=no
[home]
exten = s,1,Answer
exten = s,3,Playback(thank-you-cooperation)
exten = s,4,WaitExten
exten = _1XXX,1,Playback(thank-you-cooperation)
exten = _1XXX,2,Answer
exten = _1XXX,3,Wait(1)
exten = _1XXX,4,Playback(thank-you-for-calling)
exten = _1XXX,5,Dial(SIP/${EXTEN},10)
exten = _1XXX,8,Voicemail(u${EXTEN})
exten = _1XXX,9,Hangup
exten = _1XXX,103,Voicemail(b${EXTEN})
exten = _1XXX,104,Hangup
exten = _9.,1,Answer
exten = _9.,1,Playback(thank-you-cooperation)
exten = _9.,2,Dial(Zap/g1/${EXTEN})
[incoming]
exten = s,1,Answer()
exten = s,2,Background(/tmp/greetings)
;exten = s,2,Background(enter-phone-number10)
exten = 1,1,Playback(digits/1)
exten = 1,2,Goto(sumiya,s,1)
exten = 2,1,Playback(digits/2)
exten = 2,2,Goto(ganbaa,s,1)
exten = i,1,Playback(pbx-invalid)
exten = i,2,Goto(incoming,s,1)
exten = t,1,Playback(vm-goodbye)
exten = t,2,Hangup( )
[sumiya]
exten = s,1,Dial(SIP/1001,10)
exten = s,2,Hangup
[ganbaa]
exten = s,1,Dial(SIP/1000,10)
exten = s,2,Hangup
Regards,
Ganbaa
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Re: [asterisk-users] outgoing call problem
Hi Marco,
I forget one thing. Our zapline connected Panasonic KX-TA616 PBX.
Thanks and regards,
Ganbaa
- Original Message -
From: Marco Mouta [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, July 10, 2006 10:31 PM
Subject: Re: [asterisk-users] outgoing call problem
I'm not a a guru, but
Check this line:
exten = _9.,2,Dial(Zap/g1/${EXTEN})
do you really want to dial digit 9 through your ZapLine? are you
connected to another pbx?
If you don't want do dial 9 to PSTN line , but you want your users to
dial 9 to place outgoing calls, try this:
exten = _9.,2,Dial(Zap/g1/${EXTEN:1})
Hope it helps.
Ps. Give me some feedback if you solved the problem
On 7/10/06, Ganbaa [EMAIL PROTECTED] wrote:
Hi,
I have configured digium tdm04b card with asterisk on debian. Incoming
call
is ok. But outgoing call has problem. Would you give me advice ?
Here is my config files.
zaptel.conf
fxsks=1
fxsks=2
fxsks=3
fxsks=4
loadzone=us
defaultzone=us
zapata.conf
[channels]
language=en
context=incoming
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
transfer=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=1
txgain=4
group=1
callgroup=1
pickupgroup=1
immediate=no
musiconhold=default
busydetect=yes
callprogress=no
channel = 1-4
extension.conf
[general]
static=yes
writeprotect=no
[home]
exten = s,1,Answer
exten = s,3,Playback(thank-you-cooperation)
exten = s,4,WaitExten
exten = _1XXX,1,Playback(thank-you-cooperation)
exten = _1XXX,2,Answer
exten = _1XXX,3,Wait(1)
exten = _1XXX,4,Playback(thank-you-for-calling)
exten = _1XXX,5,Dial(SIP/${EXTEN},10)
exten = _1XXX,8,Voicemail(u${EXTEN})
exten = _1XXX,9,Hangup
exten = _1XXX,103,Voicemail(b${EXTEN})
exten = _1XXX,104,Hangup
exten = _9.,1,Answer
exten = _9.,1,Playback(thank-you-cooperation)
exten = _9.,2,Dial(Zap/g1/${EXTEN})
[incoming]
exten = s,1,Answer()
exten = s,2,Background(/tmp/greetings)
;exten = s,2,Background(enter-phone-number10)
exten = 1,1,Playback(digits/1)
exten = 1,2,Goto(sumiya,s,1)
exten = 2,1,Playback(digits/2)
exten = 2,2,Goto(ganbaa,s,1)
exten = i,1,Playback(pbx-invalid)
exten = i,2,Goto(incoming,s,1)
exten = t,1,Playback(vm-goodbye)
exten = t,2,Hangup( )
[sumiya]
exten = s,1,Dial(SIP/1001,10)
exten = s,2,Hangup
[ganbaa]
exten = s,1,Dial(SIP/1000,10)
exten = s,2,Hangup
Regards,
Ganbaa
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Marco Mouta
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[asterisk-users] outgoing call problem
Hi,
I have configured digium tdm04b card with asterisk
on debian. Incoming call is ok. But outgoing call has problem. Would you give me
advice ?
Here is my config files.
zaptel.conf
fxsks=1fxsks=2fxsks=3fxsks=4
loadzone=usdefaultzone=us
zapata.conf
[channels]language=en
context=incoming
signalling=fxs_ksusecallerid=yeshidecallerid=nocallwaiting=nousecallingpres=yescallwaitingcallerid=yestransfer=noechocancel=yesechocancelwhenbridged=yesechotraining=yesrxgain=1txgain=4group=1callgroup=1pickupgroup=1immediate=nomusiconhold=defaultbusydetect=yescallprogress=nochannel = 1-4
extension.conf
[general]static=yeswriteprotect=no
[home]
exten = s,1,Answerexten =
s,3,Playback(thank-you-cooperation)exten = s,4,WaitExten
exten =
_1XXX,1,Playback(thank-you-cooperation)exten = _1XXX,2,Answerexten
= _1XXX,3,Wait(1)exten =
_1XXX,4,Playback(thank-you-for-calling)exten =
_1XXX,5,Dial(SIP/${EXTEN},10)exten =
_1XXX,8,Voicemail(u${EXTEN})exten = _1XXX,9,Hangupexten =
_1XXX,103,Voicemail(b${EXTEN})exten = _1XXX,104,Hangup
exten = _9.,1,Answerexten =
_9.,1,Playback(thank-you-cooperation)exten =
_9.,2,Dial(Zap/g1/${EXTEN})
[incoming]
exten = s,1,Answer()exten =
s,2,Background(/tmp/greetings);exten =
s,2,Background(enter-phone-number10)exten =
1,1,Playback(digits/1)exten = 1,2,Goto(sumiya,s,1)exten =
2,1,Playback(digits/2)exten = 2,2,Goto(ganbaa,s,1)exten =
i,1,Playback(pbx-invalid)exten = i,2,Goto(incoming,s,1)exten =
t,1,Playback(vm-goodbye)exten = t,2,Hangup( )
[sumiya]exten =
s,1,Dial(SIP/1001,10)exten = s,2,Hangup
[ganbaa]exten =
s,1,Dial(SIP/1000,10)exten = s,2,Hangup
Regards,
Ganbaa
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Re: [asterisk-users] outgoing call problem
I'm not a a guru, but
Check this line:
exten = _9.,2,Dial(Zap/g1/${EXTEN})
do you really want to dial digit 9 through your ZapLine? are you
connected to another pbx?
If you don't want do dial 9 to PSTN line , but you want your users to
dial 9 to place outgoing calls, try this:
exten = _9.,2,Dial(Zap/g1/${EXTEN:1})
Hope it helps.
Ps. Give me some feedback if you solved the problem
On 7/10/06, Ganbaa [EMAIL PROTECTED] wrote:
Hi,
I have configured digium tdm04b card with asterisk on debian. Incoming call
is ok. But outgoing call has problem. Would you give me advice ?
Here is my config files.
zaptel.conf
fxsks=1
fxsks=2
fxsks=3
fxsks=4
loadzone=us
defaultzone=us
zapata.conf
[channels]
language=en
context=incoming
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
transfer=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=1
txgain=4
group=1
callgroup=1
pickupgroup=1
immediate=no
musiconhold=default
busydetect=yes
callprogress=no
channel = 1-4
extension.conf
[general]
static=yes
writeprotect=no
[home]
exten = s,1,Answer
exten = s,3,Playback(thank-you-cooperation)
exten = s,4,WaitExten
exten = _1XXX,1,Playback(thank-you-cooperation)
exten = _1XXX,2,Answer
exten = _1XXX,3,Wait(1)
exten = _1XXX,4,Playback(thank-you-for-calling)
exten = _1XXX,5,Dial(SIP/${EXTEN},10)
exten = _1XXX,8,Voicemail(u${EXTEN})
exten = _1XXX,9,Hangup
exten = _1XXX,103,Voicemail(b${EXTEN})
exten = _1XXX,104,Hangup
exten = _9.,1,Answer
exten = _9.,1,Playback(thank-you-cooperation)
exten = _9.,2,Dial(Zap/g1/${EXTEN})
[incoming]
exten = s,1,Answer()
exten = s,2,Background(/tmp/greetings)
;exten = s,2,Background(enter-phone-number10)
exten = 1,1,Playback(digits/1)
exten = 1,2,Goto(sumiya,s,1)
exten = 2,1,Playback(digits/2)
exten = 2,2,Goto(ganbaa,s,1)
exten = i,1,Playback(pbx-invalid)
exten = i,2,Goto(incoming,s,1)
exten = t,1,Playback(vm-goodbye)
exten = t,2,Hangup( )
[sumiya]
exten = s,1,Dial(SIP/1001,10)
exten = s,2,Hangup
[ganbaa]
exten = s,1,Dial(SIP/1000,10)
exten = s,2,Hangup
Regards,
Ganbaa
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Marco Mouta
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Re: [asterisk-users] outgoing call problem
Another note:
On Mon, Jul 10, 2006 at 10:05:22PM +0900, Ganbaa wrote:
Hi,
I have configured digium tdm04b card with asterisk on debian. Incoming
call is ok. But outgoing call has problem. Would you give me advice ?
It would have helped to get a CLI trace of of such a problematic call,
BTW.
See below...
Here is my config files.
zaptel.conf
fxsks=1
fxsks=2
fxsks=3
fxsks=4
loadzone=us
defaultzone=us
zapata.conf
[channels]
language=en
context=incoming
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
transfer=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=1
txgain=4
group=1
callgroup=1
pickupgroup=1
immediate=no
musiconhold=default
busydetect=yes
callprogress=no
channel = 1-4
extension.conf
[general]
static=yes
writeprotect=no
Are you sure?
[home]
exten = s,1,Answer
Missing priority 2.
exten = s,3,Playback(thank-you-cooperation)
exten = s,4,WaitExten
exten = _1XXX,1,Playback(thank-you-cooperation)
exten = _1XXX,2,Answer
exten = _1XXX,3,Wait(1)
exten = _1XXX,4,Playback(thank-you-for-calling)
exten = _1XXX,5,Dial(SIP/${EXTEN},10)
Mising priorities 6,7
exten = _1XXX,8,Voicemail(u${EXTEN})
exten = _1XXX,9,Hangup
exten = _1XXX,103,Voicemail(b${EXTEN})
exten = _1XXX,104,Hangup
exten = _9.,1,Answer
exten = _9.,1,Playback(thank-you-cooperation)
exten = _9.,2,Dial(Zap/g1/${EXTEN})
double priority 1 (try 'show dialplan home' and you'll only see the
first). Also, as already mentioned, use ${EXTEN:1}
To avoid all of those manual numbering and renumbering issues, use the
priority 'n' whereever possible:
exten = s,1,Answer
exten = s,n,Playback(thank-you-cooperation)
exten = s,n,WaitExten
--
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755 iax:[EMAIL PROTECTED]
+972-50-7952406
[EMAIL PROTECTED] http://www.xorcom.com
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[Asterisk-Users] Outgoing Call problem with PSTN line
Hi, I've got an Asterisk system that I've just added an X100P card to. Incoming calls route to my call group just fine. When I make outgoing calls by prefixing with 9, they route to the PSTN network okay, get answered but then drop straight away. Has anyone seen this before and found a fix? I'm in the UK using a phone line from NTL. I've Googled for a bit, can find others who had the same problem but they didn't post a solution. This is what I see in the console: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/2002-7743' -- Executing Macro(SIP/2002-189f, dialout|1|9123) in new stack -- Executing GotoIf(SIP/2002-189f, 1?4) in new stack -- Goto (macro-dialout,s,4) -- Executing GotoIf(SIP/2002-189f, 1?6) in new stack -- Goto (macro-dialout,s,6) -- Executing SetVar(SIP/2002-189f, length=1) in new stack -- Executing Dial(SIP/2002-189f, ZAP/g0/123) in new stack -- Called g0/123 -- Zap/1-1 answered SIP/2002-189f -- Hungup 'Zap/1-1' == Spawn extension (macro-dialout, s, 7) exited non-zero on 'SIP/2002-189f' in macro 'dialout' == Spawn extension (from-internal, 9123, 1) exited non-zero on 'SIP/2002-189f' -- Executing Macro(SIP/2002-189f, hangupcall) in new stack -- Executing ResetCDR(SIP/2002-189f, w) in new stack -- Executing NoCDR(SIP/2002-189f, ) in new stack -- Executing Wait(SIP/2002-189f, 5) in new stack -- Executing Hangup(SIP/2002-189f, ) in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/2002-189f' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/2002-189f' My zaptel.conf has: fxsks = 1 loadzone = uk defaultzone = uk Thanks, Mark ___ 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
