provided by Star2Billing S.L.

please see

This looks to be a NAT problem. There are two non-routable addresses mentioned during these dialogues: 10.11.0.6 and 192.168.1.19. You need to ensure re-invites are disabled for the peers concerned (to force the audio streams through Asterisk, so it can handle the NAT translations properly), and check you've correctly configured Asterisk's NAT settings (externip, localnets, etc).

Hello Gents,

I am having similar problem.
Card authentication works then I hear "The number you have dialed is not available at the moment.".

[May 11 15:41:50] WARNING[12015] pbx.c: The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan? (Dial(SIP/8801712043321@callwithus|60|HRrL(1333000:61000:30000)))
[May 11 15:41:50] VERBOSE[12015] netsock2.c: == Using SIP RTP TOS bits 184
[May 11 15:41:50] VERBOSE[12015] netsock2.c: == Using SIP RTP CoS mark 5
[May 11 15:41:50] ERROR[12015] netsock2.c: getaddrinfo("callwithus|60|HRrL(1333000:61000:30000)", "(null)", ...): No address associated with hostname
[May 11 15:41:50] WARNING[12015] chan_sip.c: No such host: callwithus|60|HRrL(1333000:61000:30000)
[May 11 15:41:50] WARNING[12015] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[May 11 15:41:50] VERBOSE[12015] app_dial.c: == Everyone is busy/congested at this time (1:0/0/1)
[May 11 15:41:50] VERBOSE[12015] res_agi.c: -- Playing 'prepaid-dest-unreachable' (escape_digits=#) (sample_offset 0)
[May 11 15:41:53] VERBOSE[12015] file.c: -- <SIP/87.117.75.1-00000034> Playing 'prepaid-enter-dest.gsm' (language 'en')
[May 11 15:41:55] VERBOSE[12015] res_agi.c: -- <SIP/87.117.75.1-00000034>AGI Script a2billing.php completed, returning 4
[May 11 15:41:55] VERBOSE[12015] pbx.c: == Spawn extension (a2billing, 442032981516, 2) exited non-zero on 'SIP/87.117.75.1-00000034'


Any idea?

Thanks

M

set asterisk version to 1_6

Joe