Hi...
i set up an IAX2 trunk with VoipTlak
and now iam getting..Number is currently Unavailable
i get the following error:
[Dec 8 16:04:32] VERBOSE[2421] logger.c: --- (9 headers 0 lines) ---
[Dec 8 16:04:32] VERBOSE[2421] logger.c:
<--- SIP read from
UDP://77.240.48.94:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK29d4dd35;rport=5060
From: <sip:
[email protected]>;tag=as1c47720b
To: <sip:
[email protected]>;tag=fd79486175647ed1617969929fdaad02.b0b3
Call-ID:
[email protected]CSeq: 109 REGISTER
Contact: <sip:
[email protected]>;expires=120
Server: OpenSIPS (1.5.3-notls (x86_64/linux))
Content-Length: 0
Warning: 392 77.240.48.94:5060 "Noisy feedback tells: pid=26507 req_src_ip=79.78.21.93 req_src_port=5060 in_uri=sip:voiptalk.org out_uri=sip:voiptalk.org via_cnt==1"
<------------->
[Dec 8 16:04:32] VERBOSE[2421] logger.c: --- (10 headers 0 lines) ---
[Dec 8 16:04:32] VERBOSE[2421] logger.c: Scheduling destruction of SIP dialog '
[email protected]' in 32000 ms (Method: REGISTER)
[Dec 8 16:04:32] NOTICE[2421] chan_sip.c: Outbound Registration: Expiry for voiptalk.org is 120 sec (Scheduling reregistration in 105 s)
[Dec 8 16:04:36] VERBOSE[5819] logger.c: -- Playing 'prepaid-you-have' (escape_digits=#) (sample_offset 0)
[Dec 8 16:04:37] VERBOSE[5819] logger.c: -- <SIP/4119417-b6e16e00> Playing 'digits/5.gsm' (language 'en')
[Dec 8 16:04:38] NOTICE[5819] rtp.c: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 217.10.69.13
[Dec 8 16:04:38] VERBOSE[5819] logger.c: -- Playing 'credit' (escape_digits=#) (sample_offset 0)
[Dec 8 16:04:39] VERBOSE[5819] logger.c: -- Playing 'vm-and' (escape_digits=#) (sample_offset 0)
[Dec 8 16:04:39] VERBOSE[5819] logger.c: -- <SIP/4119417-b6e16e00> Playing 'digits/70.gsm' (language 'en')
[Dec 8 16:04:40] VERBOSE[5819] logger.c: -- <SIP/4119417-b6e16e00> Playing 'digits/6.gsm' (language 'en')
[Dec 8 16:04:41] VERBOSE[5819] logger.c: -- Playing 'prepaid-cents' (escape_digits=#) (sample_offset 0)
[Dec 8 16:04:42] VERBOSE[5819] logger.c: -- <SIP/4119417-b6e16e00> Playing 'prepaid-enter-dest.gsm' (language 'en')
[Dec 8 16:04:56] VERBOSE[5819] logger.c: -- Playing 'prepaid-you-have' (escape_digits=#) (sample_offset 0)
[Dec 8 16:04:57] VERBOSE[5819] logger.c: -- <SIP/4119417-b6e16e00> Playing 'digits/1.gsm' (language 'en')
[Dec 8 16:04:58] VERBOSE[5819] logger.c: -- <SIP/4119417-b6e16e00> Playing 'digits/hundred.gsm' (language 'en')
[Dec 8 16:04:59] VERBOSE[5819] logger.c: -- Playing 'prepaid-minutes' (escape_digits=#) (sample_offset 0)
[Dec 8 16:05:00] VERBOSE[5819] logger.c: -- AGI Script Executing Application: (DIAL) Options: (IAX2/IAX2/442084712971|60|HRrL(6000000:61000:30000))
[Dec 8 16:05:00] DEBUG[5819] chan_iax2.c: prepending 8 to prefs
[Dec 8 16:05:00] VERBOSE[5819] logger.c: -- Called IAX2/442084712971|60|HRrL(6000000:61000:30000)
[Dec 8 16:05:00] WARNING[2458] chan_iax2.c: Call rejected by 217.14.138.130: No authority found
[Dec 8 16:05:00] VERBOSE[5819] logger.c: -- Hungup 'IAX2/IAX2-2705'
[Dec 8 16:05:00] VERBOSE[5819] logger.c: == Everyone is busy/congested at this time (1:0/0/1)
[Dec 8 16:05:00] VERBOSE[5819] logger.c: -- Playing 'prepaid-dest-unreachable' (escape_digits=#) (sample_offset 0)
[Dec 8 16:05:03] VERBOSE[5819] logger.c: -- <SIP/4119417-b6e16e00> Playing 'prepaid-enter-dest.gsm' (language 'en')
[Dec 8 16:05:04] VERBOSE[2421] logger.c: Really destroying SIP dialog '
[email protected]' Method: REGISTER
[Dec 8 16:05:04] VERBOSE[2421] logger.c: Really destroying SIP dialog '
[email protected]' Method: REGISTER
[Dec 8 16:05:14] VERBOSE[5819] logger.c: -- Playing 'prepaid-you-have' (escape_digits=#) (sample_offset 0)
[Dec 8 16:05:15] VERBOSE[5819] logger.c: -- <SIP/4119417-b6e16e00> Playing 'digits/1.gsm' (language 'en')
[Dec 8 16:05:16] VERBOSE[5819] logger.c: -- <SIP/4119417-b6e16e00> Playing 'digits/hundred.gsm' (language 'en')
[Dec 8 16:05:17] VERBOSE[5819] logger.c: -- Playing 'prepaid-minutes' (escape_digits=#) (sample_offset 0)
[Dec 8 16:05:17] VERBOSE[2421] logger.c: Reliably Transmitting (no NAT) to 217.10.79.23:5060:
OPTIONS sip:sipgate.co.uk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK236f53c8;rport
Max-Forwards: 70
From: "Unknown" <sip:
[email protected]>;tag=as1054ce1e
To: <sip:sipgate.co.uk>
Contact: <sip:
[email protected]>
Call-ID:
[email protected]CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Date: Tue, 08 Dec 2009 16:05:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0
any Idea? is it A2Billing not sending in right format to the Provider?
help........