outbound audio channel non working

Since more or less the beginning of the year the outbound service, that until then had worked normally, has the outbound audio channel non working.

Yes, I've used your pftlinux application to test all ports. It shows that there are no ports blocked.
Helpful to know: my Asterisk box is registered with at least four different service providers (both IAX and SIP) and with none I experience the problem that I'm having with voipuser.

Any hint on what it could be or suggestion about what to do to debug this issue?

Thanks,

--kalogero

Hi

Sorry to hear that you are having a problem.

Quote:

Any hint on what it could be or suggestion about what to do to debug this issue?

Well as with all problems like this, Not being able to any of the config file means it not realy possible to say what may be the problem. We could put lots of ideas forward, But by seeing the relevent conf files its a lot quicker and simpler.

Ian

Ian, thanks for the reply. Excerpts from Asterisk configuration files follow.

Operating details on the issue: if party 'A' is at the VoIP end and party 'B' at any PSTN end, party 'B' will hear party 'A' voice, whereas party 'A' will not hear party 'B' voice.

The data link in the direction B-->A looks absouloutly dataless after few initial seconds with some traffic.

My statement derives from observing the traffic on the Ethernet port of the ADSL modem (therefore the firewall is not involved) with the command 'iftop -i ppp0' issued on the firewall.
From this side it looks like party 'B' is not sending data after a short initial exchange.
The data link in the other direction though looks normal with a bandwidth usage of about 70kB/sec.

--kalogero

------------------------------------
ASTERISK CONFIG FILES


SIP.CONF
--------
register => <my-id>:<my-pwd>@voipuser.org

[voipuser]
type=friend
context=from-my-line
username=<my-id>
secret=<my-pwd>
host=sip.voipuser.org
fromuser=<my-id>
fromdomain=voipuser.org
insecure=very
qualify=3000
nat=yes
disallow=all
allow=alaw
allow=ulaw
musiconhold=default
language=en

EXTENSIONS.CONF
---------------
UK_VOIP_USER_TEL=<44...>
UK_VOIP_USER_IDN=<my-id>

[voipuser-out]
exten => _888.,1,macro(outgoing-line,${UK_VOIP_USER_TEL},${UK_VOIP_USER_IDN},SIP/${EXTEN:3}@voipuser)

[from-my-line]
exten => s,1,macro(my-calls-handler)


[macro-outgoing-line]
;
; Outgoing line macro
; ${ARG1} - TSP assigned telephone number
; ${ARG2} - TSP identification number/name
; ${ARG3} - Dial string specific to this TSP
;
exten => s,1,SetCallerID(${ARG1})
exten => s,n,SetCIDName(${ARG2})
exten => s,n,Dial(${ARG3})
exten => s,n,Playback(invalid)
exten => s,n,Hangup

[macro-my-calls-handler]
;
; Handler of incoming calls to my lines macro
; ${ARG1} - Not defined
;
exten => s,1,NoOp("From ---> ${CALLERID}")
exten => s,n,Wait(2)
exten => s,n,Answer()
exten => s,n,NoOp()
exten => s,n,Playback(abcd)
exten => s,n,Playback(efgl)
exten => s,n,Dial(${ALL_EXT}/s|${XFR_TMOUT}|ht,${ANSW_TMOUT},tr)
exten => s,n,Voicemail(u${INS3800_EXTN})
exten => s,n,Hangup
exten => s,102,NoOp("*** ---> 102")
exten => s,103,NoOp("*** ---> 103")
exten => s,104,NoOp("*** ---> 104")
exten => s,105,NoOp("*** ---> 105")
exten => s,106,NoOp("*** ---> 106")
exten => s,107,NoOp("*** ---> 107")
exten => s,108,Voicemail(b${INS3800_EXTN})
exten => s,109,Hangup


RTP.conf
--------
rtpstart=17700
rtpend=17800

Is there anythink I can do at my side to help you track down what's going on on the VOIPUSER server while this issue manifest itself?

--kalogero