Inbound calls not working :sip trunk: LIBERTALK

Good day Forum:

-Its been a few days ive been playing with asterisk and some sip trunks i have…as far as now im quite happy , BUT:

-One of my trunks its not getting inbound calls
-when i call , nothing happens , even with debug on
-The provider is Libertalk
-i succeded with inbound calls for a diferent provider…

PLEASE HELP ME !!!

I attach my config , log and info:

-Asterisk 11.3.0, Copyright © 1999 - 2012 Digium, Inc. and others.
-installed from source
-in VPS, publically addressable IP

########################################################################################################################
SIP.CONF
########################################################################################################################
[general]
disallow=all
allow=ulaw,alaw
qualify=yes
nat= no
bindport=5060
bindaddr=IPxxPRIVATE

register => +339904XXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org:PASSWORD:NDI04XXXXXXXX.LIBERTALK@sfr.fr@internet.p-cscf.sfr.net:5064~3600

[Neuftalk-out]
type=peer
fromdomain=ims.mnc010.mcc208.3gppnetwork.org
fromuser=+339904XXXXXXXX
defaultuser=NDI04XXXXXXXX.LIBERTALK@sfr.fr
host=internet.p-cscf.sfr.net
insecure=invite
remotesecret=PASSWORD
canreinvite=no
auth = NDI04XXXXXXXX.LIBERTALK@sfr.fr:PASSWORD@ims.mnc010.mcc208.3gppnetwork.org
outboundproxy:5064=internet.p-cscf.sfr.net
nat= no
qualify=yes
port=5064
insecure=port,invite
context=incoming_lib
permit=IPxxPRIVATE

[Neuftalk-in]
type=friend
fromdomain=ims.mnc010.mcc208.3gppnetwork.org
host=internet.p-cscf.sfr.net
port=5064
insecure=port,invite
context=incoming_lib
nat=no
qualify=yes
permit=IPxxPRIVATE

[102]
type=friend
username=102
secret=PASSWORD
host=dynamic

[103]
type=friend
username=103
secret=PASSWORD
host=dynamic

[202]
type=friend
username=202
secret=PASSWORD
host=dynamic

########################################################################################################################
extensions.conf
########################################################################################################################

[default]
exten => 102,1,Dial(SIP/102)
exten => 202,2,Dial(SIP/202)

exten => i,1,Answer
exten => +339904XXXXXXXX,2,VoiceMail(102,u)
exten => +339904XXXXXXXX,3,VoiceMail(202,u)

[incoming_lib]
exten => +339904XXXXXXXX,1,answer
exten => +339904XXXXXXXX,2,Playback(hello-world)

########################################################################################################################
extensions.conf
########################################################################################################################

[root@ftp ~]# asterisk -vvvvr
Asterisk 11.3.0, Copyright © 1999 - 2012 Digium, Inc. and others.
Created by Mark Spencer markster@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

Connected to Asterisk 11.3.0 currently running on ftp (pid = 25780)
ftpCLI> sip set debug on
SIP Debugging enabled
ftp
CLI> reload
== Parsing ‘/etc/asterisk/extconfig.conf’: Found
== Parsing ‘/etc/asterisk/logger.conf’: Found
Asterisk Queue Logger restarted
== Parsing ‘/etc/asterisk/cel.conf’: Found
– CEL logging disabled.
== Parsing ‘/etc/asterisk/codecs.conf’: Found
– Reloading module ‘app_amd.so’ (Answering Machine Detection Application)
– Reloading module ‘app_confbridge.so’ (Conference Bridge Application)
– Reloading module ‘app_followme.so’ (Find-Me/Follow-Me Application)
– Reloading module ‘app_minivm.so’ (Mini VoiceMail (A minimal Voicemail e-mail System))
– Reloading module ‘app_playback.so’ (Sound File Playback Application)
– Reloading module ‘app_queue.so’ (True Call Queueing)
[Apr 15 15:42:10] NOTICE[25912]: app_queue.c:7712 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action.
– Reloading module ‘app_voicemail.so’ (Comedian Mail (Voicemail System))
– Reloading module ‘cdr_csv.so’ (Comma Separated Values CDR Backend)
– Reloading module ‘cdr_custom.so’ (Customizable Comma Separated Values CDR Backend)
== Parsing ‘/etc/asterisk/cdr_custom.conf’: Found
– Reloading module ‘cdr_manager.so’ (Asterisk Manager Interface CDR Backend)
– Reloading module ‘cel_custom.so’ (Customizable Comma Separated Values CEL Backend)
== Parsing ‘/etc/asterisk/cel_custom.conf’: Found
– Reloading module ‘cel_manager.so’ (Asterisk Manager Interface CEL Backend)
– Reloading module ‘chan_agent.so’ (Agent Proxy Channel)
– Reloading module ‘chan_iax2.so’ (Inter Asterisk eXchange (Ver 2))
– Reloading module ‘chan_mgcp.so’ (Media Gateway Control Protocol (MGCP))
Reloading MGCP
– Reloading module ‘chan_sip.so’ (Session Initiation Protocol (SIP))
Reloading SIP
– Reloading module ‘chan_skinny.so’ (Skinny Client Control Protocol (Skinny))
[Apr 15 15:42:10] NOTICE[25912]: chan_skinny.c:7732 config_load: Configuring skinny from skinny.conf
== Parsing ‘/etc/asterisk/skinny.conf’: Found
– Reloading module ‘chan_unistim.so’ (UNISTIM Protocol (USTM))
Reloading unistim.conf…
== Parsing ‘/etc/asterisk/unistim.conf’: Found
– Reloading module ‘codec_adpcm.so’ (Adaptive Differential PCM Coder/Decoder)
– Reloading module ‘codec_alaw.so’ (A-law Coder/Decoder)
– Reloading module ‘codec_dahdi.so’ (Generic DAHDI Transcoder Codec Translator)
– Reloading module ‘codec_g722.so’ (ITU G.722-64kbps G722 Transcoder)
– Reloading module ‘codec_g726.so’ (ITU G.726-32kbps G726 Transcoder)
– Reloading module ‘codec_gsm.so’ (GSM Coder/Decoder)
– Reloading module ‘codec_lpc10.so’ (LPC10 2.4kbps Coder/Decoder)
– Reloading module ‘codec_ulaw.so’ (mu-Law Coder/Decoder)
– Reloading module ‘pbx_config.so’ (Text Extension Configuration)
== Parsing ‘/etc/asterisk/extensions.conf’: Found
– Registered extension context ‘incoming_lib’; registrar: pbx_config
– Added extension ‘0456851563’ priority 1 to incoming_lib
– Added extension ‘0456851563’ priority 2 to incoming_lib
– Registered extension context ‘default’; registrar: pbx_config
– Added extension ‘+339904XXXXXXXX’ priority 1 to default
– Added extension ‘+339904XXXXXXXX’ priority 2 to default
– Registered extension context ‘incoming’; registrar: pbx_config
– Added extension ‘2XXXXXX’ priority 1 to incoming
– Added extension ‘2XXXXXX’ priority 2 to incoming
== Parsing ‘/etc/asterisk/users.conf’: Found
– Registered extension context ‘parkedcalls’; registrar: features
– merging incls/swits/igpats from old(parkedcalls) to new(parkedcalls) context, registrar = pbx_config
– Added extension ‘700’ priority 1 to parkedcalls
– Time to scan old dialplan and merge leftovers back into the new: 0.000111 sec
– Time to restore hints and swap in new dialplan: 0.000004 sec
– Time to delete the old dialplan: 0.000005 sec
– Total time merge_contexts_delete: 0.000120 sec
– Reloading module ‘pbx_dundi.so’ (Distributed Universal Number Discovery (DUNDi))
== Parsing ‘/etc/asterisk/dundi.conf’: Found
– Reloading module ‘res_adsi.so’ (ADSI Resource)
– Reloading module ‘res_calendar.so’ (Asterisk Calendar integration)
– Reloading module ‘res_clialiases.so’ (CLI Aliases)
– Reloading module ‘res_config_sqlite3.so’ (SQLite 3 realtime config engine)
– Reloading module ‘res_crypto.so’ (Cryptographic Digital Signatures)
– Reloading module ‘res_fax.so’ (Generic FAX Applications)
– Reloading module ‘res_musiconhold.so’ (Music On Hold Resource)
– Reloading module ‘res_phoneprov.so’ (HTTP Phone Provisioning)
== Parsing ‘/etc/asterisk/sip.conf’: Found
== Parsing ‘/etc/asterisk/users.conf’: Found
== Parsing ‘/etc/asterisk/phoneprov.conf’: Found
– Reloading module ‘res_rtp_asterisk.so’ (Asterisk RTP Stack)
– Reloading module ‘res_stun_monitor.so’ (STUN Network Monitor)
Reliably Transmitting (no NAT) to 200.112.225.159:5060:
OPTIONS sip:voipserver.txi.cl SIP/2.0
Via: SIP/2.0/UDP my_server_ip:5060;branch=z9hG4bK3534d5af
Max-Forwards: 70
From: “asterisk” sip:2XXXXXX@my_server_ip;tag=as70d0a91e
To: sip:voipserver.txi.cl
Contact: sip:2XXXXXX@my_server_ip:5060
Call-ID: 108626e47b6067974e49102500b4cfc5@my_server_ip:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.3.0
Date: Mon, 15 Apr 2013 13:42:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


Reliably Transmitting (no NAT) to 91.68.1.20:5064:
OPTIONS sip:internet.p-cscf.sfr.net SIP/2.0
Via: SIP/2.0/UDP my_server_ip:5060;branch=z9hG4bK426ac955
Max-Forwards: 70
From: “asterisk” sip:+339904XXXXXXXX@my_server_ip;tag=as4e56fc06
To: sip:internet.p-cscf.sfr.net
Contact: sip:+339904XXXXXXXX@my_server_ip:5060
Call-ID: 6fde68cd42627f1500598fb073ae115b@my_server_ip:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.3.0
Date: Mon, 15 Apr 2013 13:42:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


[Apr 15 15:42:10] NOTICE[25811]: chan_sip.c:14947 sip_reregister: – Re-registration for 2XXXXXX@voipserver.txi.cl
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 200.112.225.159:5060:
REGISTER sip:voipserver.txi.cl SIP/2.0
Via: SIP/2.0/UDP my_server_ip:5060;branch=z9hG4bK720c8137
Max-Forwards: 70
From: sip:2XXXXXX@voipserver.txi.cl;tag=as4dc571e2
To: sip:2XXXXXX@voipserver.txi.cl
Call-ID: 10b380114f59113662023e646ba62e6a@my_server_ip
CSeq: 124 REGISTER
User-Agent: Asterisk PBX 11.3.0
Authorization: Digest username=“2XXXXXX”, realm=“asterisk”, algorithm=MD5, uri=“sip:voipserver.txi.cl”, nonce=“0aa72dbd”, response="825c20f990aa61a09a3e3a8e0c62bc3f"
Expires: 120
Contact: sip:2XXXXXX@my_server_ip:5060
Content-Length: 0


<— SIP read from UDP:91.68.1.20:5064 —>
SIP/2.0 407 Proxy Authentication Required
Call-ID: 6fde68cd42627f1500598fb073ae115b@my_server_ip:5060
Via: SIP/2.0/UDP my_server_ip:5060;received=my_server_ip;branch=z9hG4bK426ac955
To: sip:internet.p-cscf.sfr.net;tag=51105a36-1366026132512394
From: “asterisk” sip:+339904XXXXXXXX@my_server_ip;tag=as4e56fc06
CSeq: 102 OPTIONS
Date: Mon, 15 Apr 2013 11:42:12 GMT
Server: Alcatel-Lucent-HPSS/3.0.3
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘6fde68cd42627f1500598fb073ae115b@my_server_ip:5060’ Method: OPTIONS
[Apr 15 15:42:10] NOTICE[25811]: chan_sip.c:14947 sip_reregister: – Re-registration for +339904XXXXXXXX@internet.p-cscf.sfr.net
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 91.68.1.20:5064:
REGISTER sip:ims.mnc010.mcc208.3gppnetwork.org SIP/2.0
Via: SIP/2.0/UDP my_server_ip:5060;branch=z9hG4bK2f1259dc
Max-Forwards: 70
From: sip:+339904XXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org;tag=as2efd46b3
To: sip:+339904XXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org
Call-ID: 2ceb80111b5e74ec7b69c6ea156657f0@my_server_ip
CSeq: 108 REGISTER
User-Agent: Asterisk PBX 11.3.0
Authorization: Digest username="NDI0456851563.LIBERTALK@sfr.fr", realm=“sfr.fr”, algorithm=MD5, uri=“sip:ims.mnc010.mcc208.3gppnetwork.org”, nonce=“b7c9036dbf3054ae516be774a940e9703dc8f84c1608”, response=“d31a4747ecbd4550afd7eb866073cf82”, opaque=“ALU:QbkRBthOEgEQAkhWV1hYRAIBHgkdHwQCQ1lFRkBWGBkpdio3JnZ0ZiAnOGI-KD1-PzcnbmBmbmg_”, qop=auth, cnonce=“7cae9f4c”, nc=00000002
Expires: 3600
Contact: sip:s@my_server_ip:5060
Content-Length: 0


<— SIP read from UDP:91.68.1.20:5064 —>
SIP/2.0 401 Unauthorized
Call-ID: 2ceb80111b5e74ec7b69c6ea156657f0@my_server_ip
Via: SIP/2.0/UDP my_server_ip:5060;received=my_server_ip;branch=z9hG4bK2f1259dc
To: sip:+339904XXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org;tag=5058f170-1366026132615902
From: sip:+339904XXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org;tag=as2efd46b3
CSeq: 108 REGISTER
Date: Mon, 15 Apr 2013 11:42:12 GMT
Server: Alcatel-Lucent-HPSS/3.0.3
WWW-Authenticate: Digest realm=“sfr.fr”, nonce=“b7516be792c9036dbf3054aea940e9703dc8f84c0208”, opaque=“ALU:QbkRBthOEgEQAkhWV1hYRAIBHgkdHwQCQ1lFRkBWGBkpdio3JnZ0ZiAnOGI-KD1-PzcnbmBmbmg_”, stale=true, algorithm=MD5, qop="auth"
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Responding to challenge, registration to domain/host name internet.p-cscf.sfr.net
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 91.68.1.20:5064:
REGISTER sip:ims.mnc010.mcc208.3gppnetwork.org SIP/2.0
Via: SIP/2.0/UDP my_server_ip:5060;branch=z9hG4bK54440c3a
Max-Forwards: 70
From: sip:+339904XXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org;tag=as5a337c97
To: sip:+339904XXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org
Call-ID: 2ceb80111b5e74ec7b69c6ea156657f0@my_server_ip
CSeq: 109 REGISTER
User-Agent: Asterisk PBX 11.3.0
Authorization: Digest username="NDI0456851563.LIBERTALK@sfr.fr", realm=“sfr.fr”, algorithm=MD5, uri=“sip:ims.mnc010.mcc208.3gppnetwork.org”, nonce=“b7516be792c9036dbf3054aea940e9703dc8f84c0208”, response=“6c9441e2271c20a3655b9994f8ae9dcd”, opaque=“ALU:QbkRBthOEgEQAkhWV1hYRAIBHgkdHwQCQ1lFRkBWGBkpdio3JnZ0ZiAnOGI-KD1-PzcnbmBmbmg_”, qop=auth, cnonce=“72b5c6d5”, nc=00000001
Expires: 3600
Contact: sip:s@my_server_ip:5060
Content-Length: 0


<— SIP read from UDP:200.112.225.159:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP my_server_ip:5060;branch=z9hG4bK3534d5af;received=my_server_ip
From: “asterisk” sip:2XXXXXX@my_server_ip;tag=as70d0a91e
To: sip:voipserver.txi.cl;tag=as6e00a66e
Call-ID: 108626e47b6067974e49102500b4cfc5@my_server_ip:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:200.112.225.159
Accept: application/sdp
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Really destroying SIP dialog ‘108626e47b6067974e49102500b4cfc5@my_server_ip:5060’ Method: OPTIONS

<— SIP read from UDP:91.68.1.20:5064 —>
SIP/2.0 200 OK
Call-ID: 2ceb80111b5e74ec7b69c6ea156657f0@my_server_ip
Via: SIP/2.0/UDP my_server_ip:5060;received=my_server_ip;branch=z9hG4bK54440c3a
To: sip:+339904XXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org;tag=5058f170-1366026132639742
From: sip:+339904XXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org;tag=as5a337c97
CSeq: 109 REGISTER
Allow-Events: reg
Contact: sip:s@my_server_ip:5060;expires=3161
Date: Mon, 15 Apr 2013 11:42:12 GMT
Path: sip:pcgw-0003.imsgroup0-000.ach4isc01.ims.sfr.net:5064;lr;ottag=ue_term;bidx=2693980;access-type=ADSL
P-Associated-URI: Main sip:+339904XXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org
P-Associated-URI: Alias tel:+339904XXXXXXXX
Server: Alcatel-Lucent-HPSS/3.0.3
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Scheduling destruction of SIP dialog ‘2ceb80111b5e74ec7b69c6ea156657f0@my_server_ip’ in 32000 ms (Method: REGISTER)
[Apr 15 15:42:10] NOTICE[25811]: chan_sip.c:23275 handle_response_register: Outbound Registration: Expiry for internet.p-cscf.sfr.net is 3161 sec (Scheduling reregistration in 3146 s)

<— SIP read from UDP:200.112.225.159:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP my_server_ip:5060;branch=z9hG4bK720c8137;received=my_server_ip
From: sip:2XXXXXX@voipserver.txi.cl;tag=as4dc571e2
To: sip:2XXXXXX@voipserver.txi.cl
Call-ID: 10b380114f59113662023e646ba62e6a@my_server_ip
CSeq: 124 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------->
— (10 headers 0 lines) —

<— SIP read from UDP:200.112.225.159:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP my_server_ip:5060;branch=z9hG4bK720c8137;received=my_server_ip
From: sip:2XXXXXX@voipserver.txi.cl;tag=as4dc571e2
To: sip:2XXXXXX@voipserver.txi.cl;tag=as036c2250
Call-ID: 10b380114f59113662023e646ba62e6a@my_server_ip
CSeq: 124 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“2e3257a4”, stale=true
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Responding to challenge, registration to domain/host name voipserver.txi.cl
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 200.112.225.159:5060:
REGISTER sip:voipserver.txi.cl SIP/2.0
Via: SIP/2.0/UDP my_server_ip:5060;branch=z9hG4bK7fd5bc2a
Max-Forwards: 70
From: sip:2XXXXXX@voipserver.txi.cl;tag=as01ca6d72
To: sip:2XXXXXX@voipserver.txi.cl
Call-ID: 10b380114f59113662023e646ba62e6a@my_server_ip
CSeq: 125 REGISTER
User-Agent: Asterisk PBX 11.3.0
Authorization: Digest username=“2XXXXXX”, realm=“asterisk”, algorithm=MD5, uri=“sip:voipserver.txi.cl”, nonce=“2e3257a4”, response="1fec47bfd3dc92e7d85188fc814027a0"
Expires: 120
Contact: sip:2XXXXXX@my_server_ip:5060
Content-Length: 0


<— SIP read from UDP:200.112.225.159:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP my_server_ip:5060;branch=z9hG4bK7fd5bc2a;received=my_server_ip
From: sip:2XXXXXX@voipserver.txi.cl;tag=as01ca6d72
To: sip:2XXXXXX@voipserver.txi.cl
Call-ID: 10b380114f59113662023e646ba62e6a@my_server_ip
CSeq: 125 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------->
— (10 headers 0 lines) —

<— SIP read from UDP:200.112.225.159:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP my_server_ip:5060;branch=z9hG4bK7fd5bc2a;received=my_server_ip
From: sip:2XXXXXX@voipserver.txi.cl;tag=as01ca6d72
To: sip:2XXXXXX@voipserver.txi.cl;tag=as036c2250
Call-ID: 10b380114f59113662023e646ba62e6a@my_server_ip
CSeq: 125 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Expires: 120
Contact: sip:2XXXXXX@my_server_ip:5060;expires=120
Date: Mon, 15 Apr 2013 12:42:34 GMT
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Scheduling destruction of SIP dialog ‘10b380114f59113662023e646ba62e6a@my_server_ip’ in 32000 ms (Method: REGISTER)
[Apr 15 15:42:11] NOTICE[25811]: chan_sip.c:23275 handle_response_register: Outbound Registration: Expiry for voipserver.txi.cl is 120 sec (Scheduling reregistration in 105 s)
Really destroying SIP dialog ‘2ceb80111b5e74ec7b69c6ea156657f0@my_server_ip’ Method: REGISTER
Really destroying SIP dialog ‘10b380114f59113662023e646ba62e6a@my_server_ip’ Method: REGISTER
Really destroying SIP dialog ‘6f922e6f5435426a61485eee0cbc2277@200.112.225.159’ Method: NOTIFY

<— SIP read from UDP:200.112.225.159:5060 —>
NOTIFY sip:2XXXXXX@my_server_ip:5060 SIP/2.0
Via: SIP/2.0/UDP 200.112.225.159:5060;branch=z9hG4bK3fb25473;rport
From: “asterisk” sip:asterisk@200.112.225.159;tag=as139a53d9
To: sip:2XXXXXX@my_server_ip:5060
Contact: sip:asterisk@200.112.225.159
Call-ID: 048c40e7390014b06eb06c6f469b5700@200.112.225.159
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 96

Messages-Waiting: yes
Message-Account: sip:asterisk@200.112.225.159
Voice-Message: 9/0 (0/0)
<------------->
— (12 headers 3 lines) —

<— Transmitting (no NAT) to 200.112.225.159:5060 —>
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 200.112.225.159:5060;branch=z9hG4bK3fb25473;rport;received=200.112.225.159
From: “asterisk” sip:asterisk@200.112.225.159;tag=as139a53d9
To: sip:2XXXXXX@my_server_ip:5060;tag=as65037fd7
Call-ID: 048c40e7390014b06eb06c6f469b5700@200.112.225.159
CSeq: 102 NOTIFY
Server: Asterisk PBX 11.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>

It is difficult to work out which ITSP is which.

I only see a register for one ITSP.

permit should, normally, specify the public address of the ITSP. Generally it is of limited value on static hosts.

If you get nothing back, not even register rejections, the problem lies outside Asterisk.

I didnt post the other trunk because it works correctly :

ftp*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
voipserver.txi.cl:5060 N 24XXXXX@voip 105 Registered Mon, 15 Apr 2013 17:27:28
internet.p-cscf.sfr.net:5064 N +3399045XXXX 3193 Registered Mon, 15 Apr 2013 17:27:27

I got rid of the permits … but i get NOTHING … This trunk has been so complicated to configure !!
YIKES !!!

PLEASE GIVE ME THE POWER !!!
REQUEST ADITIONAL INFO IF NECESARy !!!

Is my_server_ip your public address?

YES IT IS !!!

For the record:

-I give up with libertalk , it may work for outgoing calls , but i couldnt make it work for incoming…
-I tested incoming calls with microsip , and it doesnt work either…

THIS CRAPPY SIP TRUNK IS CURSED !!!

i found an error in the log thought:
[Apr 16 00:00:28] WARNING[30404] loader.c: Error loading module ‘chan_dahdi.so’: libpri.so.1.4: cannot open shared object file: No such file or directory

If incoming calls work for you with libertalk(sfr) france , please post your config …

I would like to reopen this thread. I subscribed to the same operator than OP, and am kind of stuck in the same area.
I am able to place calls, but incoming calls wouldn’t succeed. Thing is, while in sip debug mode, I don’t see anything coming from the SIP trunk when calling the number. Outgoing calls work, and the operator’s proprietary application (a SIP client) does receive calls. Here’s my configuration:

sip.conf:

[general]
context=default 
bindport=5060   
register => +3399XXXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org:password:NDIXXXXXXXXXX.LIBERTALK@sfr.fr@internet.p-cscf.sfr.net:5064~3600
nat=no
localnet=192.168.1.0/255.255.255.0
externip=gateway.public.ip.address

[sfr-out]
type=peer
fromdomain=ims.mnc010.mcc208.3gppnetwork.org
fromuser=+3399XXXXXXXXXX
defaultuser=NDIXXXXXXXXXX.LIBERTALK@sfr.fr
host=internet.p-cscf.sfr.net
insecure=invite
remotesecret=password
canreinvite=no
auth=NDIXXXXXXXXXX.LIBERTALK@sfr.fr:password@ims.mnc010.mcc208.3gppnetwork.org
outboundproxy=internet.p-cscf.sfr.net:5064
nat=yes

[sfr-in]
type=friend
fromdomain=ims.mnc010.mcc208.3gppnetwork.org
host=internet.p-cscf.sfr.net
insecure=invite
context=from-sfr
port=5064
nat=yes

; phone entries follow

The “from-sfr” context is very simple :

[from-sfr]
exten => s,1,Dial(SIP/snom300) ; snom300 is defined in sip.conf

FWIW, here’s the SIP trace of the proprietary app that works for both calling and be called :

REGISTER sip:ims.mnc010.mcc208.3gppnetwork.org SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;rport;branch=z9hG4bK1784149870
From: "XXXXXXXXXX" <sip:+3399XXXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=3837542383
To: "XXXXXXXXXX" <sip:+3399XXXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>
Call-ID: 3821288860
CSeq: 1 REGISTER
Contact: <sip:+3399XXXXXXXXXX@192.168.1.1:5060;user=phone>;reg-id=1;+sip.instance="0b6ccced-67d0-4ede-8b5a-5bf395893390"
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, REFER, NOTIFY, SUBSCRIBE, REGISTER, MESSAGE
Max-Forwards: 70
User-Agent: phapi/eXosip-1.0.0
Expires: 3600
Supported: path
Supported: outbound
Content-Length: 0


SIP/2.0 401 Unauthorized
Call-ID: 3821288860
Via: SIP/2.0/UDP 192.168.1.1:5060;received=1.2.3.4;branch=z9hG4bK1784149870;rport=5060
To: "XXXXXXXXXX" <sip:+3399XXXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=506b631c-1390812949568477
From: "XXXXXXXXXX" <sip:+3399XXXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=3837542383
CSeq: 1 REGISTER
Date: Mon, 27 Jan 2014 08:55:49 GMT
Server: Alcatel-Lucent-HPSS/3.0.3
Supported: outbound
WWW-Authenticate: Digest realm="sfr.fr",
   nonce="barfoo",
   opaque="ALU:foofoo",
   algorithm=MD5,
   qop="auth"
Content-Length: 0


REGISTER sip:ims.mnc010.mcc208.3gppnetwork.org SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;rport;branch=z9hG4bK3114902737
From: "XXXXXXXXXX" <sip:+3399XXXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=3837542383
To: "XXXXXXXXXX" <sip:+3399XXXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>
Call-ID: 3821288860
CSeq: 2 REGISTER
Contact: <sip:+3399XXXXXXXXXX@192.168.1.1:5060;user=phone>;reg-id=1;+sip.instance="0b6ccced-67d0-4ede-8b5a-5bf395893390"
Authorization: Digest username="NDIXXXXXXXXXX.LIBERTALK@sfr.fr", realm="sfr.fr", nonce="b7c9036dbf3054a52e61f13ea940e9703dc8f84c1508", uri="sip:ims.mnc010.mcc208.3gppnetwork.org", response="foobar", algorithm=MD5, cnonce="0a4f113b", opaque="ALU:barfoo", qop=auth, nc=00000001
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, REFER, NOTIFY, SUBSCRIBE, REGISTER, MESSAGE
Max-Forwards: 70
User-Agent: phapi/eXosip-1.0.0
Expires: 3600
Supported: path
Supported: outbound
Content-Length: 0


SIP/2.0 403 Forbidden
Call-ID: 3821288860
Via: SIP/2.0/UDP 192.168.1.1:5060;received=1.2.3.4;branch=z9hG4bK3114902737;rport=5060
To: "XXXXXXXXXX" <sip:+3399XXXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=506b631c-1390812949621939
From: "XXXXXXXXXX" <sip:+3399XXXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=3837542383
CSeq: 2 REGISTER
Date: Mon, 27 Jan 2014 08:55:49 GMT
Server: Alcatel-Lucent-HPSS/3.0.3
Content-Length: 0


REGISTER sip:ims.mnc010.mcc208.3gppnetwork.org SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;rport;branch=z9hG4bK597152933
From: "XXXXXXXXXX" <sip:+3399XXXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=3837542383
To: "XXXXXXXXXX" <sip:+3399XXXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>
Call-ID: 3821288860
CSeq: 3 REGISTER
Contact: <sip:+3399XXXXXXXXXX@192.168.1.1:5060;user=phone>;reg-id=1;+sip.instance="0b6ccced-67d0-4ede-8b5a-5bf395893390"
Authorization: Digest username="NDIXXXXXXXXXX.LIBERTALK@sfr.fr", realm="sfr.fr", nonce="barbar", uri="sip:ims.mnc010.mcc208.3gppnetwork.org", response="15b298b762a4351f925d05e71c1c82a2", algorithm=MD5, cnonce="0a4f113b", opaque="ALU:bazbaz", qop=auth, nc=00000002
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, REFER, NOTIFY, SUBSCRIBE, REGISTER, MESSAGE
Max-Forwards: 70
User-Agent: phapi/eXosip-1.0.0
Expires: 0
Supported: path
Supported: outbound
Content-Length: 0


SIP/2.0 403 Forbidden
Call-ID: 3821288860
Via: SIP/2.0/UDP 192.168.1.1:5060;received=1.2.3.4;branch=z9hG4bK597152933;rport=5060
To: "XXXXXXXXXX" <sip:+3399XXXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=506b631c-1390812949765598
From: "XXXXXXXXXX" <sip:+3399XXXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=3837542383
CSeq: 3 REGISTER
Date: Mon, 27 Jan 2014 08:55:49 GMT
Server: Alcatel-Lucent-HPSS/3.0.3
Content-Length: 0


REGISTER sip:ims.mnc010.mcc208.3gppnetwork.org SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;rport;branch=z9hG4bK3284430795
From: "XXXXXXXXXX" <sip:+3399XXXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=338529908
To: "XXXXXXXXXX" <sip:+3399XXXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>
Call-ID: 694675161
CSeq: 1 REGISTER
Contact: <sip:+3399XXXXXXXXXX@192.168.1.1:5060;user=phone>;reg-id=1;+sip.instance="0b6ccced-67d0-4ede-8b5a-5bf395893390"
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, REFER, NOTIFY, SUBSCRIBE, REGISTER, MESSAGE
Max-Forwards: 70
User-Agent: phapi/eXosip-1.0.0
Expires: 3600
Supported: path
Supported: outbound
Content-Length: 0


SIP/2.0 401 Unauthorized
Call-ID: 694675161
Via: SIP/2.0/UDP 192.168.1.1:5060;received=1.2.3.4;branch=z9hG4bK3284430795;rport=5060
To: "XXXXXXXXXX" <sip:+3399XXXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=506b631c-1390812974111217
From: "XXXXXXXXXX" <sip:+3399XXXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=338529908
CSeq: 1 REGISTER
Date: Mon, 27 Jan 2014 08:56:14 GMT
Server: Alcatel-Lucent-HPSS/3.0.3
Supported: outbound
WWW-Authenticate: Digest realm="sfr.fr",
   nonce="zabzab",
   opaque="ALU:bazbaz",
   algorithm=MD5,
   qop="auth"
Content-Length: 0


REGISTER sip:ims.mnc010.mcc208.3gppnetwork.org SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;rport;branch=z9hG4bK3535133780
From: "XXXXXXXXXX" <sip:+3399XXXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=338529908
To: "XXXXXXXXXX" <sip:+3399XXXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>
Call-ID: 694675161
CSeq: 2 REGISTER
Contact: <sip:+3399XXXXXXXXXX@192.168.1.1:5060;user=phone>;reg-id=1;+sip.instance="0b6ccced-67d0-4ede-8b5a-5bf395893390"
Authorization: Digest username="NDIXXXXXXXXXX.LIBERTALK@sfr.fr", realm="sfr.fr", nonce="d34db33f", uri="sip:ims.mnc010.mcc208.3gppnetwork.org", response="f00f00f00", algorithm=MD5, cnonce="0a0a0a", opaque="ALU:fofo", qop=auth, nc=00000001
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, REFER, NOTIFY, SUBSCRIBE, REGISTER, MESSAGE
Max-Forwards: 70
User-Agent: phapi/eXosip-1.0.0
Expires: 3600
Supported: path
Supported: outbound
Content-Length: 0


SIP/2.0 200 OK
Call-ID: 694675161
Via: SIP/2.0/UDP 192.168.1.1:5060;received=1.2.3.4;branch=z9hG4bK3535133780;rport=5060
To: "XXXXXXXXXX" <sip:+3399XXXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=506b631c-1390812974243571
From: "XXXXXXXXXX" <sip:+3399XXXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=338529908
CSeq: 2 REGISTER
Require: outbound
Allow-Events: reg
Contact: <sip:+3399XXXXXXXXXX@192.168.1.1:5060;user=phone>;expires=3457;+sip.instance="0b6ccced-67d0-4ede-8b5a-5bf395893390";reg-id=1
Date: Mon, 27 Jan 2014 08:56:14 GMT
Path: <sip:pcgw-0007.imsgroup0-000.ach4isc06.ims.sfr.net:5064;lr;ottag=ue_term;bidx=8204832;access-type=ADSL;ob>
P-Associated-URI: Main <sip:+3399XXXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>
P-Associated-URI: Alias <tel:+3399XXXXXXXXXX>
Server: Alcatel-Lucent-HPSS/3.0.3
Content-Length: 0


....!..B..OZ.e.f.@.^
....!..B..OZ.e.f.@.^. ....2.!..B..OZ.e9.B...
....!..B.&Gi)Nk 8N.!
....!..B.&Gi)Nk 8N.!. ....2.!..B.&Gi)N......
INVITE sip:+3399XXXXXXXXXX@192.168.1.1:5060;user=phone SIP/2.0
Call-ID: LU-1390813039586678-867144@imsgroup0-000.ach4isc06.ims.sfr.net
Via: SIP/2.0/UDP 91.68.1.28:5064;branch=z9hG4bKf114ea6d67330b6d754b0dae689c5a61510fbe0b-1390813039586844;_aluscr_
Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK_001_1390813039-586702-2599905-LucentPCSF
To: <sip:+3399XXXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org;user=phone>
From: <sip:+33617630994@172.26.22.25;user=phone>;tag=510fbe0b-1390813039586681-gm-pt-lucentPCSF-011303
CSeq: 1 INVITE
Max-Forwards: 68
Content-Type: application/sdp
Accept: application/sdp
Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,INFO,REFER,UPDATE
Contact: <sip:lucentNGFS-050178@pcgw-0007.imsgroup0-000.ach4isc06.ims.sfr.net:5064>
P-Asserted-Identity: <sip:+33617630994@172.26.22.25;user=phone>
Request-Disposition: no-fork
User-Agent: Alcatel-Lucent-HPSS v3.0.3
Content-Length: 260
P-Called-Party-ID: <tel:+3399XXXXXXXXXX>
Diversion: <sip:+33173741577;cpc=ordinary@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;reason="user-busy";privacy="off";counter=1
X-LU-SkipDFC: orig
X-LU-SkipFC: orig

v=0
o=LucentPCSF 1199354056 1199354056 IN IP4 imsgroup0-000.ach4isc06.ims.sfr.net
s=-
c=IN IP4 109.3.79.92
t=0 0
m=audio 47908 RTP/AVP 18 8 101
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=maxptime:60
a=silenceSupp:off - - - -

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 91.68.1.28:5064;branch=z9hG4bKf114ea6d67330b6d754b0dae689c5a61510fbe0b-1390813039586844;_aluscr_
Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK_001_1390813039-586702-2599905-LucentPCSF
From: <sip:+33617630994@172.26.22.25;user=phone>;tag=510fbe0b-1390813039586681-gm-pt-lucentPCSF-011303
To: <sip:+3399XXXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org;user=phone>
Call-ID: LU-1390813039586678-867144@imsgroup0-000.ach4isc06.ims.sfr.net
CSeq: 1 INVITE
User-Agent: phapi/eXosip-1.0.0
Content-Length: 0


SIP/2.0 101 Dialog Establishement
Via: SIP/2.0/UDP 91.68.1.28:5064;branch=z9hG4bKf114ea6d67330b6d754b0dae689c5a61510fbe0b-1390813039586844;_aluscr_
Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK_001_1390813039-586702-2599905-LucentPCSF
From: <sip:+33617630994@172.26.22.25;user=phone>;tag=510fbe0b-1390813039586681-gm-pt-lucentPCSF-011303
To: <sip:+3399XXXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=1634442484
Call-ID: LU-1390813039586678-867144@imsgroup0-000.ach4isc06.ims.sfr.net
CSeq: 1 INVITE
Contact: <sip:+3399XXXXXXXXXX@192.168.1.1:5060>
User-Agent: phapi/eXosip-1.0.0
Content-Length: 0


SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 91.68.1.28:5064;branch=z9hG4bKf114ea6d67330b6d754b0dae689c5a61510fbe0b-1390813039586844;_aluscr_
Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK_001_1390813039-586702-2599905-LucentPCSF
From: <sip:+33617630994@172.26.22.25;user=phone>;tag=510fbe0b-1390813039586681-gm-pt-lucentPCSF-011303
To: <sip:+3399XXXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=1634442484
Call-ID: LU-1390813039586678-867144@imsgroup0-000.ach4isc06.ims.sfr.net
CSeq: 1 INVITE
Contact: <sip:+3399XXXXXXXXXX@192.168.1.1:5060>
User-Agent: phapi/eXosip-1.0.0
Content-Length: 0


SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 91.68.1.28:5064;branch=z9hG4bKf114ea6d67330b6d754b0dae689c5a61510fbe0b-1390813039586844;_aluscr_
Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK_001_1390813039-586702-2599905-LucentPCSF
From: <sip:+33617630994@172.26.22.25;user=phone>;tag=510fbe0b-1390813039586681-gm-pt-lucentPCSF-011303
To: <sip:+3399XXXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=1634442484
Call-ID: LU-1390813039586678-867144@imsgroup0-000.ach4isc06.ims.sfr.net
CSeq: 1 INVITE
Contact: <sip:+3399XXXXXXXXXX@192.168.1.1:5060;user=phone>
User-Agent: phapi/eXosip-1.0.0
Content-Length: 0


ACK sip:+3399XXXXXXXXXX@192.168.1.1:5060;user=phone SIP/2.0
Call-ID: LU-1390813039586678-867144@imsgroup0-000.ach4isc06.ims.sfr.net
Via: SIP/2.0/UDP 91.68.1.28:5064;branch=z9hG4bKf114ea6d67330b6d754b0dae689c5a61510fbe0b-1390813039586844;_aluscr_
To: <sip:+3399XXXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=1634442484
From: <sip:+33617630994@172.26.22.25;user=phone>;tag=510fbe0b-1390813039586681-gm-pt-lucentPCSF-011303
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0


....!..B...\.7...#D)
....!..B...\.7...#D). ....2.!..B...\.7..v.9.

Phone number, IP addresses and hashes have been modified to protect the innocent :smile:

Finally got the incoming calls working using the `media_address’ keyword, working configuration is available here: asterisk-france.org/showthre … #post15800