[Asterisk-Users] Outgoing SIP Problems with Asterisk and SER on same PC

[david.waugh at berlin.de]

Hello fellow asterisk people!

I have Asterisk listening on port 5061 and SER on port 5060.

Asterisk is configured as a gateway for ISDN/Analog/H323 and also SIP.

My problems are with SIP. I can make incoming calls from SIP to asterisk
and to any of the other networks, but when I try to make an outgoing
call from Asterisk to SER I see the following in Asterisk:

-- Executing Ringing("H323/ip$192.219.85.57:2712/8570", "") in new
stack
    -- Executing Dial("H323/ip$192.219.85.57:2712/8570",
"sip/280 at sip_proxy-out|20|r") in new stack
    -- Called 280 at sip_proxy-out
    -- Got SIP response 482 "Loop Detected" back from 192.219.85.57
    -- Now forwarding H323/ip$192.219.85.57:2712/8570 to
'Local/280 at sip-incoming' (thanks to SIP/sip_proxy-out-f67d)
Jul 22 20:20:25 NOTICE[29756]: chan_local.c:378 local_alloc: No such
extension/context 280 at sip-incoming creating local channel

Jul 22 20:20:25 NOTICE[29756]: app_dial.c:232 wait_for_answer: Unable to
create local channel for call forward to 'Local/280 at sip-incoming'

  == Everyone is busy/congested at this time
    -- Timeout on H323/ip$192.219.85.57:2712/8570
  == CDR updated on H323/ip$192.219.85.57:2712/8570
    -- Executing Goto("H323/ip$192.219.85.57:2712/8570", "#|1") in new
    stack -- Goto (default,#,1)
    -- Executing Playback("H323/ip$192.219.85.57:2712/8570",
"demo-thanks") in new stack
    -- Playing 'demo-thanks' (language 'en')
  == Spawn extension (default, #, 1) exited non-zero on
'H323/ip$192.219.85.57:2712/8570'

Want I want to happen is the call to go out through Asterisk - to SER
(as SER knows where the SIP extension is) - and then onto the extension
of the person to call.

In my sip.conf I have the following:

[general]
context=sip-incoming                    ; Default context for incoming
calls autocreatepeer=yes

recordhistory=yes               ; Record SIP history by default
                                ; (see sip history / sip no history)
;realm=fedcore2.eicon.com       ; Realm for digest authentication
                                ; defaults to "asterisk"
                                ; Realms MUST be globally unique
according to RFC 3261
                                ; Set this to your host name or domain
                                name
port=5061                       ; UDP Port to bind to (SIP standard port
is 5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds
to all)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound
calls
                                ; Note: Asterisk only uses the first
                                host
disallow=all                    ; First disallow all codecs
allow=ulaw                      ; Allow codecs in order of preference

register =>Asterisk:XXXXXXX at fedcore2.XXXXXXXX.com/5000

[sip_proxy-out]
type=friend                     ; we only want to call out, not be
called secret=XXXXXXXX
username=Asterisk               ; Authentication user for outbound
proxies host=fedcore2.XXXXXXX.com

In my extensions.conf I have

exten =>_5XXX,2,Dial(sip/${EXTEN:1}@sip_proxy-out,20,r)

So that dialing an extension 5XXX rings sip extension XXX.

I also the following context to catch incoming SIP calls.
[sip-incoming]
exten=>s,1,Wait,1
exten =>s,2,Goto(default,384220,1)
exten =>5000,1,Goto(default,384220,1)
exten =>_9.,1,Goto(default,${EXTEN:1},1)

Why am I unable to make outgoing SIP calls?

I have also not made any changes to my DNS SVR settings (in case I need
to???)

I have also tried to have the following alternative in my extensions.conf
It is something like this:
Asterisk extensions.conf:

 [globals]
 SERADDRESS=XXX.XXX.XXX.XXX:5060

 [context]
 exten =>_5XXX,1,Dial(SIP/${EXTEN:1}@${SERADDRESS},20,r) 

However, this then causes a message "Message to Big" to be displayed in SER.

Many thanks for your help. I am probably doing something obvious wrong!

Thanks
David