On Mon, Oct 20, 2008 at 10:37 AM, Juan Rodríguez <[EMAIL PROTECTED]> wrote: > I do not think NAT is the problem, NAT normally gives you problems like one > way audio or no registration. > Try calling the SIP/102 on other extension: > ;TEST > exten => 1002,1,Dial(SIP,102|20) > exten => 1002,n,Hangup() > instead of: > > exten => 102,1,Dial... > But this is a very strange error... Check if there is no other definition of > default having 102 on it because Asterisk is going to merge the extensions.
I get the following when trying to dial 1002 from 101. I've attached
my extensions.conf file in-case there is something else that is
conflicting as you mentioned.
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/101-082aca90",
"SIP/102/20") in new stack
== Using SIP RTP CoS mark 5
-- Called 102/20
[Oct 20 11:03:33] WARNING[20285]: chan_sip.c:2787 retrans_pkt: Maximum
retries exceeded on transmission
[EMAIL PROTECTED] for seqno 102
(Critical Request) -- See doc/sip-retransmit.txt.
[Oct 20 11:03:33] WARNING[20285]: chan_sip.c:2814 retrans_pkt: Hanging
up call [EMAIL PROTECTED] - no reply to
our critical packet (see doc/sip-retransmit.txt).
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [EMAIL PROTECTED]:2] Hangup("SIP/101-082aca90", "") in new
stack
== Spawn extension (default, 1002, 2) exited non-zero on 'SIP/101-082aca90'
extensions.conf
Description: Binary data
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