[Asterisk-Users] Problems with * and IAXTel/FWD

Lee Redmayne lee.redmayne at nwva.org
Thu Oct 23 03:22:48 MST 2003


Hi all

I've been trying to make * work with IAXtel to no avail, all seems ok in
the config but am not getting anywhere

This is what I'm getting from console (user/pass/dest # changed for
obvious reasons):

DEBUG[1133735216]: File chan_sip.c, Line 3841 (check_user): Setting NAT
on RTP to 0
DEBUG[1133735216]: File chan_sip.c, Line 4891 (handle_request): Check
for res for phone1
DEBUG[1133735216]: File chan_sip.c, Line 973 (find_user): Call from user
'phone1' is 1 out of 0
DEBUG[1133735216]: File chan_sip.c, Line 3307 (build_route):
build_route: Contact hop: <sip:phone1 at 10.1.2.24:5060;line=1>
    -- Executing Dial("SIP/phone1-2c71",
"IAX/user:secretpass/BYEXTENSION at iaxtel") in new stack
    -- Calling using options 'exten=18007692511;callerid=phone1
<7201>;language=en;context=iaxtel;username=user;formats=2;capability=654
07;version=1;adsicpe=2'
    -- Called user:secretpass at iaxtel.com/18005551212 at iaxtel
WARNING[1125342512]: File chan_iax.c, Line 1110 (attempt_transmit): Max
retries exceeded to host 12.37.165.130 on IAX[12.37.165.130:5036]/7
(type = 6, subclass = 1, ts=1, seqno=0)
DEBUG[1209269552]: File chan_iax.c, Line 1687 (iax_hangup): We're
hanging up IAX[12.37.165.130:5036]/7 now...
    -- Hungup 'IAX[12.37.165.130:5036]/7'
  == No one is available to answer at this time
WARNING[1209269552]: File pbx.c, Line 1810 (ast_pbx_run): Timeout, but
no rule 't' in context 'sip'
DEBUG[1209269552]: File chan_sip.c, Line 1025 (sip_hangup):
find_user(phone1) - decrement inUse counter
DEBUG[1133735216]: File chan_sip.c, Line 548 (__sip_ack): Stopping
retransmission on '3c29efbbc5b1-diw483wrl88j at 10-1-2-24' of Response 1:
Found

On FWD I get the following

DEBUG[1133735216]: File chan_sip.c, Line 3841 (check_user): Setting NAT
on RTP to 0
DEBUG[1133735216]: File chan_sip.c, Line 4891 (handle_request): Check
for res for phone1
DEBUG[1133735216]: File chan_sip.c, Line 973 (find_user): Call from user
'phone1' is 1 out of 0
DEBUG[1133735216]: File chan_sip.c, Line 3307 (build_route):
build_route: Contact hop: <sip:phone1 at 10.1.2.24:5060;line=1>
    -- Executing Dial("SIP/phone1-3efc", "SIP/613 at fwd.pulver.com") in
new stack
DEBUG[1209269552]: File chan_sip.c, Line 857 (sip_call): Outgoing Call
for 613
DEBUG[1209269552]: File chan_sip.c, Line 952 (find_user): 613 is not a
local user
    -- Called 613 at fwd.pulver.com
DEBUG[1133735216]: File chan_sip.c, Line 657 (create_addr): Setting NAT
on RTP to 0
DEBUG[1133735216]: File chan_sip.c, Line 548 (__sip_ack): Stopping
retransmission on '0a21f240725f604b75d2cb3801005394 at 10.1.2.1' of Request
102: Found
WARNING[1133735216]: File chan_sip.c, Line 445 (retrans_pkt): Maximum
retries exceeded on call 7b2990bf7dee8d8551372a337b172754 at 10.1.2.1 for
seqno 102 (Request)
DEBUG[1209269552]: File chan_sip.c, Line 1022 (sip_hangup):
find_user(613) - decrement outUse counter
DEBUG[1209269552]: File chan_sip.c, Line 952 (find_user): 613 is not a
local user
  == No one is available to answer at this time
DEBUG[1133735216]: File chan_sip.c, Line 657 (create_addr): Setting NAT
on RTP to 0
DEBUG[1133735216]: File chan_sip.c, Line 548 (__sip_ack): Stopping
retransmission on '0cc64763203232c6193becae17c125ab at 10.1.2.1' of Request
102: Found
WARNING[1133735216]: File chan_sip.c, Line 445 (retrans_pkt): Maximum
retries exceeded on call 7b2990bf7dee8d8551372a337b172754 at 10.1.2.1 for
seqno 102 (Request)
WARNING[1209269552]: File pbx.c, Line 1810 (ast_pbx_run): Timeout, but
no rule 't' in context 'sip'
DEBUG[1209269552]: File chan_sip.c, Line 1025 (sip_hangup):
find_user(phone1) - decrement inUse counter
DEBUG[1133735216]: File chan_sip.c, Line 548 (__sip_ack): Stopping
retransmission on '3c2a07519978-01foxxqyi62v at 10-1-2-24' of Response 1:
Found

Any advice you can give will help enormously!

Many thanks




More information about the asterisk-users mailing list