[Asterisk-Users] Using Zyxel Analog Telephone adapter with a GSM
gateway
Stig Thune
stig.thune at telecoms-resources.no
Thu Jan 20 08:41:02 MST 2005
Searching through wiki and google.
http://www.2n.cz/products/gsm_gateways/voip/voiceblue.html
but there are also other products on the market.
---
Wondering if its possible to connect as follows:
Extension -> Asterisk -> ZyxelAnalogTelephoneAdapter -> GSM gateway.
The best way would be to make the ZyxelAnalog.. to be a channel.
But I don't think that is doable.. or ?
----
So i checked with Dial command.. trying to use something like:
exten => 99,1,Dial(SIP/11,20,D($EXTEN),w=800ms)
; Dial option
; 'D([digits])' -- Send DTMF digit string *after* called party has answered
; but before the bridge. (w=500ms sec pause)
Problem is, that the astrisk won't push the $EXTEN numbers.
(or does it ? I can't hear anything :-| )
My console (verbose level 3):
-- Executing Dial("SIP/03-031e", "SIP/11|20|D(987654321)|w=200ms") in new stack
-- Called 11
-- SIP/11-1644 is ringing
-- SIP/11-1644 answered SIP/03-031e
-- Attempting native bridge of SIP/03-031e and SIP/11-1644
Jan 20 16:27:43 WARNING[10949]: chan_sip.c:1820 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)
Jan 20 16:27:43 WARNING[10949]: chan_sip.c:1820 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)
...and so on for 10 lines.. then I hang up.
== Spawn extension (wx3trunk, 99, 1) exited non-zero on 'SIP/03-031e'
/ Stig Henning
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