Asterisk 1.8 LTS incompatibilities with v.1.4

Asterisk Asterisk Support
1

Hello community,
I have a wholesale server running Asterisk 1.8.4.4 where everything is working without any problem, being billed with a2billing 1.9.4.
The problem which I’m finding to exchange incoming traffic with other customers who’re using asterisk 1.4…

When a caller from asterisk 1.4 is calling through my server, I’m rebooting him the calls with the following trace:

[quote][Jul 4 12:44:24] WARNING[15279]: pbx.c:3846 pbx_substitute_variables_helper_full: Error in extension logic (missing ‘}’)
[Jul 4 12:44:24] WARNING[15279]: pbx.c:3476 func_args: Can’t find trailing parenthesis for function ‘CALLERID(al’?
[Jul 4 12:44:24] ERROR[15279]: func_callerid.c:984 callerid_read: Unknown callerid data type ‘al’.
– Executing [00XXXXXXXX@a2billing:1] Verbose(“SIP/XXXX-00000010”, "1,a2billing - 00XXXXXXXX - ") in new stack
a2billing - 00XXXXXXXX -
– Executing [00XXXXXXXX@a2billing:2] Wait(“SIP/XXXXX-00000010”, “1”) in new stack
– Executing [00XXXXXXXX@a2billing:3] AGI(“SIP/88924-00000010”, “a2billing.php,2”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
– AGI Script Executing Application: (DIAL) Options: (SIP/TrunkC/XXXXX#00XXXXXXXX,60,LIW(18795000:60000:30000))
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called TrunkC/XXXX#00XXXXXXXX
– SIP/TrunkC-00000011 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– <SIP/88924-00000010>AGI Script a2billing.php completed, returning -1[/quote]

Why I’m having this behavior? it’s happening only when the caller is under v.1.4…

Anyone can explain me what’s this in the earth may mean??

2

The first three lines mean you have minor coding errors in the dialplan.

The last line means that the other side side said that the destination number was busy.

Without more context, and probably a SIP trace, there is not a lot more to say, other than that you should fix your parenthesiation and the parameters of the caller ID function, to avoid future confusion from those errors.

3

[quote=“david55”]The first three lines mean you have minor coding errors in the dialplan.

The last line means that the other side side said that the destination number was busy.

Without more context, and probably a SIP trace, there is not a lot more to say, other than that you should fix your parenthesiation and the parameters of the caller ID function, to avoid future confusion from those errors.[/quote]

Thanks David for your prompt,
the dialplan, it’s true, it was my mistake that i didn’t close the synthax, but in dead this error still there, and I just discover it, but cannot explain it…

as I’m using a2billing for voip traffic, and asterisk is being running as realtime server using a2billing DB. The error deeply it’s coming from the DB. where the sip user name is being modified, mysteriously, by itself, and change the value to “s”, so in the sip registry i see the sip user/s, so when the client call, my server cannot authenticate him, cause it’s sip user name is being s, but it’s sip from user it’s correct, lastly the a2billing agi cannot write the cdr, cause the sip user name is incorrect… finally the call is being rebooted…

This error indeed is not a compatibility error as i though, and I open this thread mistaking, so i just come here to close this thread, because I just submit other one into a2billing forum here: forum.asterisk2billing.org/viewt … 454#p34454, as I think, it’s related to a2billing DB management, asterisk is not writing in the DB, it’s just reading it as a real time server… so it’s not asterisk error…

Regards,