provided by Star2Billing S.L.

Hello all,
firstly a lot of thanks for every body here becuase i lernt a lot from your tutorials and articles, So.
i have a 2trunks which is working fine under the Elastix machine 2.0.4 and i produce calling card with A2Billing 1.8 and here is the problem 1.8 works fine with one trunk but the other one dont want to run correctly -:( so when i make a call through the callplane and rate cards which using this trunk i get the following:
15:46:14 VERBOSE [25898] res_agi.c:

-- Playing 'prepaid-you-have' (escape_digits=#) (sample_offset 0)
Aug 16 15:46:15 VERBOSE [25898] file.c: -- <SIP/goe_did1-000002dd> Playing 'digits/15.gsm' (language 'en')
Aug 16 15:46:16 VERBOSE [25898] res_agi.c: -- Playing 'euros' (escape_digits=#) (sample_offset 0)
Aug 16 15:46:17 VERBOSE [25898] res_agi.c: -- Playing 'vm-and' (escape_digits=#) (sample_offset 0)
Aug 16 15:46:18 VERBOSE [25898] file.c: -- <SIP/goe_did1-000002dd> Playing 'digits/40.gsm' (language 'en')
Aug 16 15:46:18 VERBOSE [25898] file.c: -- <SIP/goe_did1-000002dd> Playing 'digits/5.gsm' (language 'en')
Aug 16 15:46:19 VERBOSE [25898] res_agi.c: -- Playing 'prepaid-cents' (escape_digits=#) (sample_offset 0)
Aug 16 15:46:20 VERBOSE [25898] file.c: -- <SIP/goe_did1-000002dd> Playing 'prepaid-enter-dest.gsm' (language 'en')
Aug 16 15:46:35 VERBOSE [25898] res_agi.c: -- AGI Script Executing Application: (DIAL) Options: (SIP/voip_info/00966534455119|60|HRrL(8428000:61000:30000))
Aug 16 15:46:35 WARNING [25898] pbx.c: The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan? (Dial(SIP/voip_info/00966534455119|60|HRrL(8428000:61000:30000)))
Aug 16 15:46:35 VERBOSE [25898] netsock2.c: == Using SIP RTP TOS bits 184
Aug 16 15:46:35 VERBOSE [25898] netsock2.c: == Using SIP RTP CoS mark 5
Aug 16 15:46:35 VERBOSE [25898] app_dial.c: -- Called SIP/voip_info/00966534455119|60|HRrL(8428000:61000:30000)
Aug 16 15:47:07 VERBOSE [25898] app_dial.c: -- SIP/voip_info-000002de is circuit-busy
Aug 16 15:47:07 VERBOSE [25898] app_dial.c: == Everyone is busy/congested at this time (1:0/1/0)
Aug 16 15:47:07 WARNING [2921] chan_sip.c: Retransmission timeout reached on transmission 0b4775e6404321ff31ae188d28b5f3e8@myip:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/ ... nsmissions Packet timed out after 32000ms with no response
Aug 16 15:47:07 VERBOSE [25898] res_agi.c:-- AGI Script Executing Application: (DIAL) Options: (SIP/voip_info/00966534455119|60|HRrL(8428000:61000:30000))
Aug 16 15:47:07 WARNING [25898] pbx.c: The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan? (Dial(SIP/voip_info/00966534455119|60|HRrL(8428000:61000:30000)))
Aug 16 15:47:07 VERBOSE [25898] netsock2.c: == Using SIP RTP TOS bits 184
Aug 16 15:47:07 VERBOSE [25898] netsock2.c: == Using SIP RTP CoS mark 5
Aug 16 15:47:07 VERBOSE [25898] app_dial.c: -- Called SIP/voip_info/00966534455119|60|HRrL(8428000:61000:30000)
Aug 16 15:47:39 VERBOSE [25898] app_dial.c: -- SIP/voip_info-000002df is circuit-busy
Aug 16 15:47:39 VERBOSE [25898] app_dial.c: == Everyone is busy/congested at this time (1:0/1/0)
Aug 16 15:47:40 VERBOSE [25898] res_agi.c: -- Playing 'prepaid-dest-unreachable' (escape_digits=#) (sample_offset 0)
Aug 16 15:47:40 WARNING [2921] chan_sip.c: Retransmission timeout reached on transmission 47290535478af21805f62b5d5e2bd544@myip:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/ ... nsmissions Packet timed out after 32125ms with no response

and finally i here "the number your call is currently not a vailable".
1- my trunk is well defined and i can make calls through elastix.
2- the same method i used i a2billing is working for another trunk which is (direct SIP Gateway).

i dont know where i am doing something wrong ... see please urgent help, because my system in production phase. thanks in advance

Hi

This comment in your logs creates a useful pointer




Did you remember to set the Asterisk version to 1_6 in the system settings?

Joe

my dear jroper,

you made my day it is working now i changed only the asterisk_ver from 1_4 to 1_6 and now ...it is running like Ferari .... -:) thanks a lot man.

Dear jroper,

Thanks for your help, i appriciat it. now i have another problem, i can register SIP/AIX freind and they can make calls also the problem is there is a lot of nice but i am using the same trunk wich give clear voice when i call with DID i dont know where is the problem. i ma using Zoiper for IP Calls. is it codecs problem? or configuration problem? my extinsions_a2billing.conf look like this:
[a2billing]
exten => _X.,1,Answer
exten => _X.,n,Wait(1)
exten => _X.,n,deadAGI(a2billing.php,1)
exten => _X.,n,Hangup

[a2billing-callback]
exten => _X.,1,deadAGI(a2billing.php,1,callback)
exten => _X.,n,Hangup

[a2billing-cid-callback]
exten => _X.,1,deadAGI(a2billing.php,1,cid-callback,34) ;last #parameter is the callback area code
exten => _X.,n,Hangup

[a2billing-all-callback]
exten => _X.,1,deadAGI(a2billing.php,1,all-callback,34) ;last #parameter is the callback area code
exten => _X.,n,Hangup

[a2billing-did]
exten => _X.,1,Goto(a2billing,${EXTEN},1)
exten => _X.,2,Hangup

[a2billing-voucher]
exten => _X.,1,deadAGI(a2billing.php,1,voucher)
exten => _X.,n,Hangup

[custom-a2billing-did]
exten => _X.,1,Goto(a2billing,${EXTEN},1)
exten => _X.,2,Hangup

[custom-a2billing]
exten => _X.,1,deadAGI(a2billing.php,1)
exten => _X.,n,Hangup

Zoiper give me when he start a call U-law? and it is also in the allowed codecs for my SIP/AIAX friend??