[Asterisk-Users] Re: Problems with incoming calls on a E1 ISDN PRI

Roberto Reiner Uhry reineruhry at gmail.com
Tue Apr 19 16:14:47 MST 2005


Hi,

More information!

Look for the last line, this occurs when I receive a phone call from
my zaptel information.

Apr 19 20:08:21 DEBUG[4878]: Set channel Zap/31-1 to read format gsm
Apr 19 20:08:21 DEBUG[4878]: Set channel SIP/6633-115a to write format gsm
Apr 19 20:08:21 DEBUG[4878]: Set channel Zap/31-1 to write format slin
Apr 19 20:08:21 DEBUG[4878]: Set channel SIP/6633-115a to read format slin
Apr 19 20:08:21 DEBUG[4878]: Ooh, format changed from unknown to gsm

On SJPhone shows a message with PCMA
Does anybody have more information?

Tkz,
Reiner
On 4/19/05, Roberto Reiner Uhry <reineruhry at gmail.com> wrote:
> Hi,
> 
> I have an Asterisk installed on a FC3 with a Digium e100p card and an
> E1 (ISDN PRI).
> I'm in Brazil and using Embratel as carrier.
> 
> After few troubles I get it working to make calls, from a SIP channel
> to an Fone through the carrier.  But when I receive a call, this one
> is transfered to the SIP channel but when I answere this one stay
> quiet.
> 
> Does anybody have any ideai about how can I solve this problem?
> 
> /etc/zaptel.conf
> loadzone = us
> defaultzone = us
> span=1,1,0,ccs,hdb3,crc4
> bchan=1-15,17-31
> dchan=16
> alaw=1-31 --- I also tried with/without this line
> 
> /etc/asterisk/extensions.conf
> [from-sip-external]
> exten => _.,1,AbsoluteTimeout(15)
> exten => _.,2,Congestion
> exten => _.,3,Hangup
> 
> [from-sip]
> exten => _66XX,1,Dial(SIP/${EXTEN},20,t)
> exten => _66XX,2,Hangup()
> exten => _0.,1,Dial(Zap/g1/${EXTEN:1})
> 
> [default]
> exten => 6689,1,Dial(SIP/6689,20,t)
> exten => 6689,2,Hangup()
> 
> /etc/asterisk/sip.conf
> [general]
> 
> port = 5060           ; Port to bind to (SIP is 5060)
> bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
> disallow=all
> allow=alaw
> allow=gsm
> context = from-sip-external ; Send unknown SIP callers to this context
> callerid = Unknown
> 
> ;include sip_nat.conf
> ;include sip_additional.conf
> 
> [6689]
> context = from-sip
> username = 6689
> secret = 6689
> host = dynamic
> type = friend
> regexten = 6689
> allow=ulaw
> 
> /etc/asterisk/zapata.conf
> [trunkgroups]
> trunkgroup => 1,16
> spanmap => 1,1,1
> 
> [channels]
> switchtype=euroisdn
> signalling=pri_cpe
> language=us
> defaultzone=us
> group = 1
> musiconhold = default
> echocancel=yes
> channel => 1-15,17-31
>



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