asterisk users - Aug 2006 - No path to translate from SIP/30-09df8928(4) to SIP/3470075XXXX-09dfb518(256)

I run unrder CentOs 4.3, and have asterisk, asterisk-addons, 
asterisk-sound, zaptel, zapata and libpri installed.
I defined so far 2 accounts;  sipgate works fine for incoming and 
outgoing calls.

The 2. account (peoplecall) makes problems:
- The "register" statement works find and quering in asterisk tells me
I
am registered
sip.peoplecall.com:5060 34700754XXXX 105 Registered
sipgate.de:5060 5550873 105 Registered

- I am able to make a call to the point that he other person picks-up 
the phone. In this very same moment, the call breaks and I see on my 
phone screen: "DECLINED <phone nbr>
- In asterisk CL I see the following:
-- Executing Dial("SIP/30-09dfbdb8",
"SIP/611111111@3470075XXXX|45|r")
in new stack
-- Called 611111111@3470075XXXX
-- SIP/3470075XXXX-09e01778 is making progress passing it to SIP/30-09dfbdb8
-- SIP/3470075XXXX-09e01778 answered SIP/30-09dfbdb8
Aug 11 08:00:24 WARNING[2612]: channel.c:2706 
ast_channel_make_compatible: No path to translate from 
SIP/30-09dfbdb8(4) to SIP/3470075XXXX-09e01778(256)
Aug 11 08:00:24 WARNING[2612]: app_dial.c:1595 dial_exec_full: Had to 
drop call because I couldn't make SIP/30-09dfbdb8 compatible with 
SIP/3470075XXXX-09e01778
== Spawn extension (default, *1611111111, 1) exited non-zero on 
'SIP/30-09dfbdb8'


My configuration files loos as follows:
sip.conf
register => 3470075XXXX001:passwd@sip.peoplecall.com/001

[3470075XXXX]
type=peer
host=sip.peoplecall.com
fromuser=3470075XXXX001
fromdomain=sip.peoplecall.com
username=3470075XXXX001
secret=passwd
dtmfmode=rfc2833
qualify=yes
disallow=all
allow=g729
allow=g723
insecure=very
nat=yes

[peoplecall_in]
type=peer
fromdomain=sip.peoplecall.com
host=sip.peoplecall.com
context=ankommend

extension.conf
exten => 5550873,1,Dial,SIP/30|30|r
exten => 5550873,2,Goto,r-${DIALSTATUS}|1

exten => 001,1,Dial,SIP/30|30|r
exten => 001,2,Goto,r-${DIALSTATUS}|1


Thanks for your help

Wolfgang Paul Rauchholz wrote:
> exten => 5550873,1,Dial,SIP/30|30|r
> exten => 001,1,Dial,SIP/30|30|r


|r is evil - Don't use it.


I would be willing to bet large sums of cash this problem will go away 
if you simply remove the |r (and reload)




Jeremy McNamara

On Fri, August 11, 2006 4:26 pm, Wolfgang Paul Rauchholz
said:
> allow=g729
> allow=g723
Do you have the g729 and g723 codecs installed? They are not installed
with Asterisk by default.

cYa,
Avi

On Friday 11 August 2006 18:26, Wolfgang Paul Rauchholz
wrote:
> Aug 11 08:00:24 WARNING[2612]: channel.c:2706
> ast_channel_make_compatible: No path to translate from
> SIP/30-09dfbdb8(4) to SIP/3470075XXXX-09e01778(256)
> Aug 11 08:00:24 WARNING[2612]: app_dial.c:1595 dial_exec_full: Had to
> drop call because I couldn't make SIP/30-09dfbdb8 compatible with
You don't have the g729 codec installed by the looks.

hads

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