[Asterisk-Users] outgoing call (Sip phones to PSTN)

[Ron S]

Hi All,

   Everytime I make outgoing call, the channel at TDM
card doing hungup after might be a second the
destination number get ringing.The call is from sip
phones to PSTN phone. The sip phones was completely
registered to asterisk. here is my conf :
sip.conf :
[1234]
type=friend
username=1234
secret=
host=dynamic
context=sip-ph

extensions.conf :
[sip-ph]
exten = _NXX,1,Dial(Zap/g1/${EXTEN}|10,t)

zapata.conf:
signalling=fxs_ks
group=1
context=incoming
channel=3-4

I thought in this case the context sip-ph doesn't
need to exist in zapata.conf,does it ?
So where is my mistakes ? Please help

thanks,
ron



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[Asterisk-Users] Asterisk recognize GSM CLI

[Ron S]

Hi All,
   Does Asterisk has the ability to recognize the CLI
that comes in GSM format, as we know that PSTN has
different format compared to GSM. Is there any
specific configuration needed ?

Thanks,
ron



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[Asterisk-Users] incoming outgoing call

[Ron S]

Hi All,
   I've installed digium TDM02B. The PSTN line
connected to this card. At the IP network side, I have
SIP phones registered sucessfully to asterisk server.
How do I configure asterisk, so once there is incoming
call to the TDM channel from PSTN, the caller will
hear another dial tone from asterisk then to key in
the extension of intended destination (SIP phones
number) that already connected behind asterisk ? I
utilize exten = s,1,dial,SIP/${EXTEN},but it doesn't
work. Appreciate for the helps.

Regards,
ron



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