Hi all,
I have a E100P digium card installed in my server, and now I am
working with the incoming call to this PRI interface.
my extensions.conf file for incoming call is:
[default]
exten => , 1,Answer
exten => s,1,Dial(SIP/2001,20)
exten => s,2,Hangup
And when the incoming call comes, it appears:
-- Executing Dial("Zap/1-1", "SIP/2001|20") in new stack
-- Called 2001
-- Accepting call from '' to '' on channel 0/1, span 1
-- SIP/2001-5ef8 is ringing
-- SIP/2001-5ef8 answered Zap/1-1
-- Channel 0/1, span 1 got hangup
== Spawn extension (default, s, 1) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
My question is what's the meaning of "Accepting call from '' to
'' on
channel 0/1, span 1", and why I must have something like "exten =>
,
1,Answer" in the 'default' context?
Since if I remove the "exten => , 1,Answer"
[default]
exten => s,1,Dial(SIP/2001,20)
exten => s,2,Hangup
It appears some error in the asterisk
-- Extension '' in context 'default' from '' does not
exist.
Rejecting call on channel 0/1, span 1
Any idea will be highly appreciated!!
Thank you in advance!!
BR,
ning
On December 15, 2004 11:13 am, Ning Zhou wrote:> my extensions.conf file for incoming call is: > [default] > exten => , 1,Answer > exten => s,1,Dial(SIP/2001,20) > exten => s,2,Hangup > > And when the incoming call comes, it appears: > > -- Executing Dial("Zap/1-1", "SIP/2001|20") in new stack > -- Called 2001 > -- Accepting call from '' to '' on channel 0/1, span 1 > -- SIP/2001-5ef8 is ringing > -- SIP/2001-5ef8 answered Zap/1-1 > -- Channel 0/1, span 1 got hangup > == Spawn extension (default, s, 1) exited non-zero on 'Zap/1-1' > -- Hungup 'Zap/1-1' > > My question is what's the meaning of "Accepting call from '' to '' on > channel 0/1, span 1", and why I must have something like "exten => , > 1,Answer" in the 'default' context?You've got a PRI that doesn't pass any digits for the DID called to you? My PRI passes me 7 digits and my dialplan looks like this: exten -> 2922001,1,Dial(IAX2/user@user2001) exten -> 2922002,1,Dial(IAX2/user@user2002) and then a catchall: exten -> _29220XX,1,Answer exten -> _29220XX,2,Play(num-i-have) exten -> _29220XX,3,SayDigits(${EXTEN}) exten -> _29220XX,4,Play(goodbye) exten -> _29220XX,5,Hangup Are you positive that your telco isn't sending you any digits? -A.
hi all I have a x100p install in my server to conect my sip extensions whith a PABX. When i call from a sip phone to a pstn number it work well, but to call from pstn line to asterisk only can do it to especific phone doing this: exten => s,1,Dial(SIP/116) the problem is that i want to call any of my sip extensions not one directly. thanks in advance wert __________________________________ Do you Yahoo!? Yahoo! Mail - now with 250MB free storage. Learn more. http://info.mail.yahoo.com/mail_250
There is no way for asterisk to know which extension you want to call when you have a single analog line. You can either send the call to an extension or group, or you could create a menu system that allow the caller to select which extension to call (press 1 for Steve, 2 for Dave etc.). /Anders> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of R A > Sent: den 15 december 2004 21:20 > To: Ning Zhou; Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Incoming call problem > > hi all > > I have a x100p install in my server to conect my sip > extensions whith a PABX. When i call from a sip phone to a > pstn number it work well, but to call from pstn line to > asterisk only can do it to especific phone doing this: > > exten => s,1,Dial(SIP/116) > > > the problem is that i want to call any of my sip extensions > not one directly. > > thanks in advance > > wert > > > > > __________________________________ > Do you Yahoo!? > Yahoo! Mail - now with 250MB free storage. Learn more. > http://info.mail.yahoo.com/mail_250 > _______________________________________________ > 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 >