asterisk users - Jan 2005 - Outgoing SIP call from Asterisk problem

Hello,

I'm having a problem I can't seen to figure out.  In a nut shell, I have
asterisk running with 4 accounts configured.  All accounts work fine for 
  local calling to each other and voicemail.  However, only 1 account 
can make outgoing calls.  All the others fail with the following error.

If anyone can shed some light on the possible problem or where to look 
for more info it would be greatly appreciated.

     -- Accepting AUTHENTICATED call from 172.xx.xx.xx, requested format 
= 1024, actual format = 1024
     -- Executing Dial("IAX2/xxxx@xxxx/1", 
"SIP/00441223xxxxxx@voipuser|40|r") in new stack
     -- Called 00441223xxxxxx@voipuser
Jan 18 11:19:21 WARNING[27087]: chan_sip.c:6811 handle_response: 
Forbidden - wrong password on authentication for INVITE to '""xxxx
xxxx"
<sip:xxxxxxxx@217.xxx.xxx.xxx>;tag=as6471df4d'
     -- SIP/voipuser-d3ba is circuit-busy
   == Everyone is busy/congested at this time
     -- Executing Congestion("IAX2/xxxx@xxxx/1", "2") in new
stack
     -- Got SIP response 483 "Too Many Hops" back from 216.127.66.119
   == Spawn extension (local, *99*441223xxxxxx, 2) exited non-zero on 
'IAX2/yuki@yuki/1'
Jan 18 11:19:23 ERROR[1588]: cdr_addon_mysql.c:122 mysql_log: cdr_mysql: 
cannot connect to database server xxxxxxxxx.  Call will not be logged
     -- Hungup 'IAX2/xxxx@xxxx/1'

The Mysql error is another error if anyone has any input on that :)  The 
user/passwd and database all exist and work fine, the table for the 
database has been created and is valid.

Thanks in advance.  I have been searching for days for an answer to this 
  with no success at all.

Cheers!!!


-- 
Robert P. McKenzie                 |   GammaRay Technical Services Ltd
rmckenzi@rpmdp.com                 |             rob@gammaray-tech.com
http://www.uk-experience.com       |      http://www.gammaray-tech.com

Ecademy Profile:   http://www.ecademy.com/account.php?op=view&id=64014

Hello ALL

This is bending my head and I'm hoping someone can help.  Call flow as
follows

Sipphone -> sipphone.com -> asterisk -> sipura 3000 (PSTN port)

the sipphone is calling to my pstn line on the sipura.  Works FANTASTIC, pin
number on sipura entered, DTMF to PSTN decoded and number dialed, no problems !

IN the configurations below, the supra cant decode the DTMF, nor can the
remote asterisk box decode the extension properly for the PSTN gateway


mobile -> ipkall.com -> sipphone.com -> asterisk -> sipura 3000
mobile -> sipura 3000 -> asterisk -> sipphone.com -> asterisk ->
sipura
sipphone -> asterisk -> sipphone.com -> asterisk -> sipura

Using asterisk -vvvvc I see broken extension numbers ie: if it was extension
7010, it decodes 73, 0 701 etc

With a straight sipphone in the top example, it works 100% reliable.  If
anyone could help I'd surely appreciate it as I'm out of ideas !

Regards..Pete

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