[Asterisk-Users] Re: Simple SIP X-Lite Configuration Failing

[J Poz]

Sorry for any confusion.........But in my latest error, instead of calling my clients "jay" and "jtest", I'm calling them "400" and "410".. Everything else is still the same and it's same problem. 

My guess is that I've set a parameter incorrectly and therefore Asterisk thinks there's only one client so any calls I try to make between the two fail since it thinks the other client is busy. But I don't understand enough to interpret the error message. I thought the SIP part would be the easy part - I already have the FXO and FXS interfaces working.
 
Again, thanks for anyone who can help me since I am at a loss!

J Poz <jpoz0000 at yahoo.com> wrote:
Can anyone help. I've changed the extensions.conf file as follows:
 
extensions.conf
[sip] ; context for X-Lite Clients
exten =>11,1,Dial(SIP/jay,20,tr)
exten =>22,1,Dial(SIP/jtest,20,tr)
 
I'm still getting the Auto-congesting error (and circuit-busy). Does anyone know what is causing this in such a simple configuration?


localhost*CLI>
    -- Executing Dial("SIP/400-c3de", "SIP/410|20|tr") in new stack
    -- Called 410
May  2 19:15:56 NOTICE[1133742896]: chan_sip.c:1021 auto_congest: Auto-congesting SIP/410-a4a1
    -- SIP/410-a4a1 is circuit-busy
  == Everyone is busy at this time

J Poz <jpoz0000 at yahoo.com> wrote:
I keep getting the following Auto-congesting message whenever I try to dial from an X-Lite SIP phone to another one within my LAN. It's a real basic configuration but I am unable to figure out what is happening:
 
  localhost*CLI>
    -- Executing Dial("SIP/jay-de1b", "SIP/jtest|20|tr") in new stack
    -- Called jtest
May  2 11:47:58 NOTICE[1133742896]: chan_sip.c:1019 auto_congest: Auto-congestin
g SIP/jtest-6a1e
    -- SIP/jtest-6a1e is circuit-busy
  == Everyone is busy at this time
May  2 11:47:58 WARNING[1226062640]: pbx.c:1198 pbx_extension_helper: No applica
tion 'DialCongestion' for extension (sip, 22, 2)
  == Spawn extension (sip, 22, 2) exited non-zero on 'SIP/jay-de1b'
 
My setup is very simple and basic:
SIP.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = sip; Default

[jay]
type=friend
secret=jaysip
auth=md5
nat=yes
host=dynamic
reinvite=no
canreinvite=no
qualify=100
dtmfmode=inband
callerid="Jay <400>"
disallow=all
allow=gsm
context=sip

[jtest]
type=friend
secret=jaytestsip
auth=md5
nat=yes
host=dynamic
reinvite=no
canreinvite=no
qualify=100
dtmfmode=inband
callerid="Jay Test <410
disallow=all
allow=gsm
context=sip
 
extensions.conf
[sip] ; context for X-Lite Clients
exten =>11,1,Dial(SIP/jay,20,tr)
exten =>11,2,Congestion
exten =>22,1,Dial(SIP/jtest,20,tr)
exten =>22,2,DialCongestion
 
Lastly, here's my client setup
Display Name: Jay
User Name & Authorization User: jay
Password: jaysip 
Domain/Realm: 192.168.1.20 
SIP Proxy: 192.168.1.20
 
Display Name: Jay Test
User Name & Authorization User: jtest
Password: jaytestsip 
Domain/Realm: 192.168.1.20 
SIP Proxy: 192.168.1.20

 
Any help anyone can give me would be appreciated since I've already spent HOURS on this and have made absolutely no progress in debugging this (didn't find anything in any of the archives nor wiki pages).
 
J... 




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