[Asterisk-Users] Fwd: Sip calling errors

[iMRAN]

---------- Forwarded message ----------
From: iMRAN <ronny.net at gmail.com>
Date: May 1, 2005 12:16 PM
Subject: Sip calling errors
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>, Alexander Scheerschmidt
<alexander.scheerschmidt at telenet.be>


Hi Pros,

I`m new to Asterisk Getting following errors on my * :

  -- Executing Dial("SIP/1000-ee7c", "SIP/19166889297 at venus") in new stack
  -- Called 19166889297 at venus
Apr 28 21:06:09 WARNING[2268]: channel.c:2115
ast_channel_make_compatible: No path to translate from
SIP/venus-e8ba(2) to SIP/1000-ee7c(256)
Apr 28 21:06:09 NOTICE[2268]: pbx.c:1318 pbx_extension_helper: Cannot
find extension context 'default'
Apr 28 21:06:09 NOTICE[2268]: channel.c:1724 ast_set_read_format:
Unable to find a path from g729 to gsm
Apr 28 21:06:09 NOTICE[2268]: channel.c:1691 ast_set_write_format:
Unable to find a path from gsm to g729
  -- SIP/venus-e8ba is making progress passing it to SIP/1000-ee7c
RFC3389: 1 bytes, level 256...
Apr 28 21:06:10 NOTICE[2268]: rtp.c:298 process_rfc3389: RFC3389
support incomplete.  Turn off on client if possible
Apr 28 21:06:13 NOTICE[2268]: pbx.c:1318 pbx_extension_helper: Cannot
find extension context 'default'
  -- SIP/venus-e8ba answered SIP/1000-ee7c
  -- Attempting native bridge of SIP/1000-ee7c and SIP/venus-e8ba
Apr 28 21:06:21 WARNING[2268]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call 109143024302GnHe-1000--0019166889297 at 1.1.1.1
for seqno 25090 (Non-critical Response)
Apr 28 21:06:22 WARNING[2268]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call 109143024302GnHe-1000--0019166889297 at 1.1.1.1
for seqno 25090 (Non-critical Response)onse)

========================================================
My SIP.CONF

[general]
port = 5060
bindaddr = 0.0.0.0
canreinvite=no
disallow=all
allow=g723.1
allow=alaw
allow=ulaw
allow=gsm
allow=g729

[venus]
type=friend
context=sip-dial
host=2.2.2.2
canreinvite=no
qualify=yes
disallow=all
allow=gsm
allow=g729
insecure=very
dtmfmode=info
tos=0x18

[9999]
type=friend
host=dynamic
username=9999
secret=imran
dtmf=inband
context=internal
dtmfmode=rfc2833

[1000]
type=friend
username=1000
;secret=password1
host=dynamic
allow=g729
allow=g723.1
context=internal
dtmfmode=rfc2833
=========================================

[general]
static=yes
writeprotect=yes

[globals]
PHONE1=SIP/9999
PHONE2=SIP/1000
PHONE3=SIP/1001

[internal]
include => local-sip

[local-sip]
exten => 9999,1,Dial(${PHONE1},40,t)
exten => 9999,2,Hangup

exten => 1000,1,Dial(${PHONE2},40,t)
exten => 1000,2,Hangup

exten => 1001,1,Dial(${PHONE3},40,t)
exten => 1001,2,Hangup

exten => _00.,1,Dial(SIP/${EXTEN:2}@venus)
exten => _00.,2,Hangup

Venus is my SIP provider (sorry u might hav guessed already)

1000 and 1001 belongs to my AudioCodec MP108 8 FXS SIP device and 9999
is my softphone SJphone, i can dial soft to hard and vise versa, i can
call to US number thru my SIP provider using my Sjphone (crapy sound)
but when i try to dial from MP108 i get the above errors i mentioned.

MP108 have preloaded codec i.e. g729 and g723.1, my provider supports
g729 and g723.1

please can anyone help me ?