Re: [Asterisk-Users] Asterisk Internal sip calls I can´t send/recive

[Omar Lopez Limonta]

> > solo cambia tu extension.conf
> >
> > [entrada]
> > exten => s,1,Wait,11
> > exten => s,2,Answer
> > exten => s,3,Wait,1
> > exten => s,4,Dial(SIP/200,60,Ttr)
> > exten => s,5,Dial(SIP/201,60,Ttr)
> > exten => s,6,Dial(SIP/202,60,Ttr)
> > exten => s,7,Dial(SIP/203,60,Ttr)
> >

I try it , but it doesn´t work , i want call to another sip extension
into my lan, i want call to 201 from 203 extension  both are into my
LAN in the same range using SIP Software Phone , when i call to any
extension i get
----------
ERROR
----------
Verbosity is at least 6
   -- Remote UNIX connection
   -- Executing Dial("SIP/201-979d", "SIP/201|60|t") in new stack
   -- Called 201
May 29 18:09:28 WARNING[6082]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call 0afcadc1422e63800115943201a885fb at 192.168.1.44
for seqno 102 (Critical Request)
 == No one is available to answer at this time
   -- Executing VoiceMail("SIP/201-979d", "201") in new stack
   -- Playing 'vm-intro' (language 'es')
 == Spawn extension (anurix, 201, 2) exited non-zero on 'SIP/201-979d'
May 29 18:09:34 WARNING[6082]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call
C15A57EC-51A0-4157-BCE5-B09C0A99FD26 at 192.168.1.33 for seqno 52991
(Non-critical Response)
---------

And voicemail bring me on , and i can stablish SIP conection.