Hello every body,

i have elastix 2.0.4 and a2billing 1.8 running very good and i produce cards very fine. i have a problem with a trunk wich is working fine when i make a call through elastix but when i use the trunk through the call card i become a mssg" the number you are dailing is currently not a vailable" debug for the call show me this:
Aug 13 13:37:37 VERBOSE [12717] pbx.c:

-- Executing [_X.@a2billing-did:1] Goto("SIP/goe_did1-000000ab", "a2billing,_X.,1") in new stack
Aug 13 13:37:37 VERBOSE [12717] pbx.c:
-- Goto (a2billing,_X.,1)
Aug 13 13:37:37 VERBOSE [12717] pbx.c:
-- Executing [_X.@a2billing:1] Answer("SIP/goe_did1-000000ab", "") in new stack
Aug 13 13:37:37 VERBOSE [12717] pbx.c:
-- Executing [_X.@a2billing:2] Wait("SIP/goe_did1-000000ab", "1") in new stack
Aug 13 13:37:37 NOTICE [12717] channel.c:
Dropping incompatible voice frame on SIP/goe_did1-000000ab of format ulaw since our native format has changed to 0x8 (alaw)
Aug 13 13:37:38 VERBOSE [12717] pbx.c:
-- Executing [_X.@a2billing:3] DeadAGI("SIP/goe_did1-000000ab", "a2billing.php,1") in new stack
Aug 13 13:37:38 WARNING [12717] res_agi.c:
DeadAGI has been deprecated, please use AGI in all cases!
Aug 13 13:37:38 VERBOSE [12717] res_agi.c:
-- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
Aug 13 13:37:38 VERBOSE [12717] res_agi.c:
-- Playing 'prepaid-you-have' (escape_digits=#) (sample_offset 0)
Aug 13 13:37:39 VERBOSE [12717] file.c:
-- <SIP/goe_did1-000000ab> Playing 'digits/19.gsm' (language 'en')
Aug 13 13:37:40 VERBOSE [12717] res_agi.c:
-- Playing 'euros' (escape_digits=#) (sample_offset 0)
Aug 13 13:37:41 VERBOSE [12717] res_agi.c:
-- Playing 'vm-and' (escape_digits=#) (sample_offset 0)
Aug 13 13:37:42 VERBOSE [12717] file.c:
-- <SIP/goe_did1-000000ab> Playing 'digits/19.gsm' (language 'en')
Aug 13 13:37:43 VERBOSE [12717] res_agi.c:
-- Playing 'prepaid-cents' (escape_digits=#) (sample_offset 0)
Aug 13 13:37:43 VERBOSE [12717] file.c:
-- <SIP/goe_did1-000000ab> Playing 'prepaid-enter-dest.gsm' (language 'en')
Aug 13 13:38:00 VERBOSE [12717] res_agi.c:
-- AGI Script Executing Application: (DIAL) Options: (SIP/a2billing_voip/00966534455119|60|HRrL(12792000:61000:30000)00966534455119,)
Aug 13 13:38:00 VERBOSE [12717] netsock2.c:
== Using SIP RTP TOS bits 184
Aug 13 13:38:00 VERBOSE [12717] netsock2.c:
== Using SIP RTP CoS mark 5
Aug 13 13:38:00 VERBOSE [12717] app_dial.c:
-- Called SIP/a2billing_voip/00966534455119|60|HRrL(12792000:61000:30000)00966534455119
Aug 13 13:38:06 NOTICE [2921] chan_sip.c:
Aug 13 13:38:32 WARNING [2921] chan_sip.c:
Retransmission timeout reached on transmission 3780d487301385bc3e7e19ee4694c864@62.75.216.33:5060 for seqno 102 (Critical Request) -- See wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response
Aug 13 13:38:32 WARNING [2921] chan_sip.c:
Hanging up call 3780d487301385bc3e7e19ee4694c864@62.75.216.33:5060 - no reply to our critical packet (see wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
Aug 13 13:38:32 VERBOSE [12717] app_dial.c:
== Everyone is busy/congested at this time (1:0/0/1)
Aug 13 13:38:33 VERBOSE [12717] res_agi.c:
-- AGI Script Executing Application: (DIAL) Options: (SIP/a2billing_voip/00966534455119|60|HRrL(12792000:61000:30000))
Aug 13 13:38:33 WARNING [12717] pbx.c:
The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan? (Dial(SIP/a2billing_voip/00966534455119|60|HRrL(12792000:61000:30000)))
Aug 13 13:38:33 VERBOSE [12717] netsock2.c:
== Using SIP RTP TOS bits 184
Aug 13 13:38:33 VERBOSE [12717] netsock2.c:
== Using SIP RTP CoS mark 5
Aug 13 13:38:33 VERBOSE [12717] app_dial.c:
-- Called SIP/a2billing_voip/00966534455119|60|HRrL(12792000:61000:30000)
Aug 13 13:39:05 VERBOSE [12717] app_dial.c:
-- SIP/a2billing_voip-000000ad is circuit-busy


so the simulator of this call is fine and show me the right results and i have another trunk under a2billing and it makes calls very well ? so any body have an idea where i can start?


thanks

Hello,

so i think this will only for the moderator of this board, now i made a sip show "trunk" : and i get:


* Name : trunk_name
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : from-trunk-sip-trunk_name
Subscr.Cont. : <Not set>
Language : en
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
MOH Suggest :
Mailbox :
VM Extension : *97
LastMsgsSent : 32767/65535
Call limit : 0
Max forwards : 0
Dynamic : No
Callerid : "" <>
MaxCallBR : 384 kbps
Expire : -1
Insecure : no
Force rport : Yes
ACL : No
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: -1
DirectMedia : Yes
PromiscRedir : No
User=Phone : No
Video Support: Yes
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost : host_ip
Addr->IP : host_ip:port
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: user_name
SIP Options : (none)
Codecs : 0x90f (g723|gsm|ulaw|alaw|g726|g729)
Codec Order : (ulaw:20,gsm:20,alaw:20,g723:30,g726:20,g729:20)
Auto-Framing : No
100 on REG : Yes
Status : Unmonitored
Useragent :
Reg. Contact :
Qualify Freq : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No

so can any body tell me why this trunk work with elastix ext. but not with the A2Billing calling Card? i defined it with the same name in a2billing and put it as provider_ip in a2b. ?? HEEEEEEEEEEEELp i will get crazy.

Paste the output for:
asterisk -rx "sip show peers"
and also try to dial 00966534455119 from any extension and paste the CLI output (asterisk -r)