Internal Sip Call hangs up directly after dailing

Hi,

I’m quite new to Asterisk, but I have a good feeling about it.

I have setup Asterisk + A2Billing for my usage and I’m able to register a 2 Softphones on 2 seperate computers.

While testing and figuring out my settings I see that I’m not able to dail to my internal SIP account using the friends prefix, where I have set everything well for using the A2billing docs and comparing to the configuration of Asterisk, it seems to be OK.

What happens and what I only can get out of the CLI is the following. I was in no way able to display “Verbose Messages” with any option that shoud do that trick.

Here is my output so far:

Mar 2 03:06:08 DEBUG[7835]: chan_sip.c:7217 check_user_full: Setting NAT on RTP to 0
Mar 2 03:06:08 DEBUG[7835]: chan_sip.c:1412 __sip_ack: Stopping retransmission on ‘N2I1NjE0NjY3NDlkOWY5NmI4MDJiYWUxMTE3NjEwMmU.’ of Response 1: Match Found
Mar 2 03:06:08 DEBUG[7835]: chan_sip.c:7217 check_user_full: Setting NAT on RTP to 0
Mar 2 03:06:08 DEBUG[7835]: chan_sip.c:10573 handle_request_invite: Checking SIP call limits for device 6355817464
Mar 2 03:06:08 DEBUG[7835]: chan_sip.c:6196 build_route: build_route: Contact hop: sip:6355817464@192.168.1.201:5060
– Executing Answer(“SIP/6355817464-b5e01990”, “”) in new stack
– Executing Wait(“SIP/6355817464-b5e01990”, “2”) in new stack
Mar 2 03:06:08 DEBUG[7835]: chan_sip.c:1412 __sip_ack: Stopping retransmission on ‘N2I1NjE0NjY3NDlkOWY5NmI4MDJiYWUxMTE3NjEwMmU.’ of Response 2: Match Found
– Executing DeadAGI(“SIP/6355817464-b5e01990”, “a2billing.php”) in new stack
– Launched AGI Script /usr/share/asterisk/agi-bin/a2billing.php
– AGI Script a2billing.php completed, returning 0
– Executing Wait(“SIP/6355817464-b5e01990”, “2”) in new stack
– Executing Hangup(“SIP/6355817464-b5e01990”, “”) in new stack
== Spawn extension (a2billing, 9987654321, 5) exited non-zero on 'SIP/6355817464-b5e01990’
Mar 2 03:06:12 DEBUG[7984]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '294521790315398’
Mar 2 03:06:12 DEBUG[7984]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '294521790315398’
Mar 2 03:06:12 DEBUG[7984]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '9987654321’
Mar 2 03:06:12 DEBUG[7984]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'a2billing’
Mar 2 03:06:12 DEBUG[7984]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'SIP/6355817464-b5e01990’
Mar 2 03:06:12 DEBUG[7984]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '(null)'
Mar 2 03:06:12 DEBUG[7984]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'Hangup’
Mar 2 03:06:12 DEBUG[7984]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '(null)'
Mar 2 03:06:12 DEBUG[7984]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2008-03-02 03:06:08’
Mar 2 03:06:12 DEBUG[7984]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2008-03-02 03:06:08’
Mar 2 03:06:12 DEBUG[7984]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2008-03-02 03:06:12’
Mar 2 03:06:12 DEBUG[7984]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '4’
Mar 2 03:06:12 DEBUG[7984]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '4’
Mar 2 03:06:12 DEBUG[7984]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'ANSWERED’
Mar 2 03:06:12 DEBUG[7984]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'BILLING’
Mar 2 03:06:12 DEBUG[7984]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '6355817464’
Mar 2 03:06:12 DEBUG[7984]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '1204423568.5’
Mar 2 03:06:12 DEBUG[7984]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '(null)'
Mar 2 03:06:12 DEBUG[7984]: chan_sip.c:2435 sip_hangup: update_call_counter(6355817464) - decrement call limit counter
Mar 2 03:06:12 DEBUG[7835]: chan_sip.c:1412 __sip_ack: Stopping retransmission on ‘N2I1NjE0NjY3NDlkOWY5NmI4MDJiYWUxMTE3NjEwMmU.’ of Request 102: Match Found

I have pasted it without code tags because this output here is nicer :smile:

Because A2Billing does make normal Asterisk settings for calling, as far as I can see, I hope we can figure this problem out.

Thanks.

OK, solved by upgrading from 1.2.3 to the latest build.