CARRIER REGISTRATION | TRUNK BY AUTHENTICATION

[disepa]

Hi everybody,

It's been hard for me because I read any manual on vicidial but my IP trunk by authebtication (CALLR) does not registred.

This is my settings :

[CALLR]
disallow=all
allow=g729
allow=alaw
type=peer
host=xx.sip.callr.xx
dtmfmode=rfc2833
context=default
qualify=yes
nat=force_rport
insecure=port,invite

When I do a SIP show peers :

vicibox10*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
2003/2003 192.168.88.202 D Yes Yes 52916 OK (3 ms)
CALLR XX.YY.ZZ.AAA Yes Yes 5060 UNREACHABLE
KCALL DD.CCC.BBB.YY Yes No 5060 UNREACHABLE
gs102/gs102 (Unspecified) D Yes Yes 0 UNKNOWN

[disepa]

And when I do a core reload, I get this :

vicibox10*CLI> core reload
[Nov 24 23:41:06] NOTICE[9077]: cdr.c:4494 cdr_toggle_runtime_options: CDR simple logging enabled.
[Nov 24 23:41:06] -- CEL logging disabled.
[Nov 24 23:41:06] == TLS/SSL certificate ok
[Nov 24 23:41:06] Asterisk Queue Logger restarted
[Nov 24 23:41:06] -- Reloading module 'res_statsd.so' (StatsD client support)
[Nov 24 23:41:06] -- Reloading module 'res_pjproject.so' (PJPROJECT Log and Utility Support)
[Nov 24 23:41:06] -- Reloading module 'res_pjsip.so' (Basic SIP resource)
[Nov 24 23:41:06] NOTICE[10713]: sorcery.c:1333 sorcery_object_load: Type 'system' is not reloadable, maintaining previous values
[Nov 24 23:41:06] -- Reloading module 'res_xmpp.so' (Asterisk XMPP Interface)
[Nov 24 23:41:06] -- Reloading module 'res_stun_monitor.so' (STUN Network Monitor)
[Nov 24 23:41:06] -- Reloading module 'res_crypto.so' (Cryptographic Digital Signatures)
[Nov 24 23:41:06] -- Reloading module 'res_phoneprov.so' (HTTP Phone Provisioning)
[Nov 24 23:41:06] WARNING[9077]: res_phoneprov.c:1233 get_defaults: Unable to find a valid server address or name.
[Nov 24 23:41:06] -- Reloading module 'res_pjsip_outbound_publish.so' (PJSIP Outbound Publish Support)
[Nov 24 23:41:06] -- Reloading module 'res_hep.so' (HEPv3 API)
[Nov 24 23:41:06] -- Reloading module 'res_ari.so' (Asterisk RESTful Interface)
[Nov 24 23:41:06] -- Reloading module 'res_parking.so' (Call Parking Resource)
[Nov 24 23:41:06] -- Reloading module 'res_config_curl.so' (Realtime Curl configuration)
[Nov 24 23:41:06] -- Reloading module 'res_config_ldap.so' (LDAP realtime interface)
[Nov 24 23:41:06] NOTICE[9077]: res_config_ldap.c:1832 parse_config: No directory user found, anonymous binding as default.
[Nov 24 23:41:06] ERROR[9077]: res_config_ldap.c:1858 parse_config: No directory URL or host found.
[Nov 24 23:41:06] NOTICE[9077]: res_config_ldap.c:1776 reload: Cannot reload LDAP RealTime driver.
[Nov 24 23:41:06] -- Reloading module 'res_config_sqlite3.so' (SQLite 3 realtime config engine)
[Nov 24 23:41:06] -- Reloading module 'res_pjsip_authenticator_digest.so' (PJSIP authentication resource)
[Nov 24 23:41:06] -- Reloading module 'res_pjsip_endpoint_identifier_ip.so' (PJSIP IP endpoint identifier)
[Nov 24 23:41:06] -- Reloading module 'res_musiconhold.so' (Music On Hold Resource)
[Nov 24 23:41:06] -- Reloading module 'res_rtp_asterisk.so' (Asterisk RTP Stack)
[Nov 24 23:41:06] -- Reloading module 'res_pjsip_mwi.so' (PJSIP MWI resource)
[Nov 24 23:41:06] -- Reloading module 'res_pjsip_publish_asterisk.so' (PJSIP Asterisk Event PUBLISH Support)
[Nov 24 23:41:06] -- Reloading module 'chan_mgcp.so' (Media Gateway Control Protocol (MGCP))
[Nov 24 23:41:06] -- Reloading module 'chan_sip.so' (Session Initiation Protocol (SIP))
[Nov 24 23:41:06] -- Reloading module 'chan_iax2.so' (Inter Asterisk eXchange (Ver 2))
[Nov 24 23:41:06] -- Reloading module 'chan_motif.so' (Motif Jingle Channel Driver)
[Nov 24 23:41:06] -- Reloading module 'chan_dahdi.so' (DAHDI Telephony w/PRI & SS7 & MFC/R2)
[Nov 24 23:41:06] -- Reloading module 'res_pjsip_outbound_registration.so' (PJSIP Outbound Registration Support)
[Nov 24 23:41:06] -- Reloading module 'res_pjsip_notify.so' (CLI/AMI PJSIP NOTIFY Support)
[Nov 24 23:41:06] -- Reloading module 'res_pjsip_phoneprov_provider.so' (PJSIP Phoneprov Provider)
[Nov 24 23:41:06] -- Reloading module 'res_adsi.so' (ADSI Resource)
[Nov 24 23:41:06] -- Reloading module 'app_meetme.so' (MeetMe conference bridge)
[Nov 24 23:41:06] -- Reloading module 'app_confbridge.so' (Conference Bridge Application)
[Nov 24 23:41:06] -- Reloading module 'app_agent_pool.so' (Call center agent pool applications)
[Nov 24 23:41:06] -- Reloading module 'cel_manager.so' (Asterisk Manager Interface CEL Backend)
[Nov 24 23:41:06] -- Reloading module 'cel_custom.so' (Customizable Comma Separated Values CEL Backend)
[Nov 24 23:41:06] NOTICE[9077]: cel_custom.c:97 load_config: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs.
[Nov 24 23:41:06] Added CEL CSV mapping for 0 files.
[Nov 24 23:41:06] -- Reloading module 'codec_speex.so' (Speex Coder/Decoder)
[Nov 24 23:41:06] -- Reloading module 'app_amd.so' (Answering Machine Detection Application)
[Nov 24 23:41:06] -- Reloading module 'res_clialiases.so' (CLI Aliases)
[Nov 24 23:41:06] -- Reloading module 'app_voicemail.so' (Comedian Mail (Voicemail System))
[Nov 24 23:41:06] -- Reloading module 'app_osplookup.so' (Open Settlement Protocol Applications)
[Nov 24 23:41:06] -- Reloading module 'pbx_config.so' (Text Extension Configuration)
[Nov 24 23:41:06] == Setting global variable 'CONSOLE' to 'Console/dsp'
[Nov 24 23:41:06] == Setting global variable 'TRUNK' to 'DAHDI/r1'
[Nov 24 23:41:06] == Setting global variable 'TRUNKX' to 'DAHDI/r2'
[Nov 24 23:41:06] == Setting global variable 'TRUNKIAX' to 'IAX2/ASTtest1:test@10.10.10.16:4569'
[Nov 24 23:41:06] == Setting global variable 'TRUNKIAX1' to 'IAX2/ASTtest1:test@10.10.10.16:4569'
[Nov 24 23:41:06] == Setting global variable 'TRUNKBINFONE' to 'IAX2/1112223333:PASSWORD@iax.binfone.com'
[Nov 24 23:41:06] == Setting global variable 'SIPtrunk' to 'SIP/1234:PASSWORD@sip.provider.net'
[Nov 24 23:41:06] == Setting global variable 'TRUNKloop' to 'IAX2/ASTloop:BN2Pj8YPG28aThk@127.0.0.1:40569'
[Nov 24 23:41:06] == Setting global variable 'TRUNKblind' to 'IAX2/ASTblind:BN2Pj8YPG28aThk@127.0.0.1:41569'
[Nov 24 23:41:06] == Setting global variable 'TRUNKplay' to 'IAX2/ASTplay:BN2Pj8YPG28aThk@127.0.0.1:42569'
[Nov 24 23:41:06] == Setting global variable 'SIPCALLR' to 'SIP/CALLR'
[Nov 24 23:41:06] == Setting global variable 'KCALL' to 'SIP/KCALL'
[Nov 24 23:41:06] WARNING[9077]: pbx.c:7071 add_priority: Unable to register extension '_33XXXXXXXXX' priority 1 in 'vicidial-auto-external', already in use
[Nov 24 23:41:06] WARNING[9077]: pbx_config.c:1857 pbx_load_config: Unable to register extension at line 145 of /etc/asterisk/extensions-vicidial.conf
[Nov 24 23:41:06] WARNING[9077]: pbx.c:7071 add_priority: Unable to register extension '_33XXXXXXXXX' priority 2 in 'vicidial-auto-external', already in use
[Nov 24 23:41:06] WARNING[9077]: pbx_config.c:1857 pbx_load_config: Unable to register extension at line 146 of /etc/asterisk/extensions-vicidial.conf
[Nov 24 23:41:06] WARNING[9077]: pbx.c:7071 add_priority: Unable to register extension '_33XXXXXXXXX' priority 3 in 'vicidial-auto-external', already in use
[Nov 24 23:41:06] WARNING[9077]: pbx_config.c:1857 pbx_load_config: Unable to register extension at line 147 of /etc/asterisk/extensions-vicidial.conf
[Nov 24 23:41:06] Reloading SIP
[Nov 24 23:41:06] Reloading MGCP
[Nov 24 23:41:06] -- Time to scan old dialplan and merge leftovers back into the new: 0.000085 sec
[Nov 24 23:41:06] -- Time to restore hints and swap in new dialplan: 0.000002 sec
[Nov 24 23:41:06] -- Time to delete the old dialplan: 0.000056 sec
[Nov 24 23:41:06] -- Total time merge_contexts_delete: 0.000143 sec
[Nov 24 23:41:06] -- pbx_config successfully loaded 15 contexts (enable debug for details).
[Nov 24 23:41:06] -- Reloading module 'app_playback.so' (Sound File Playback Application)
[Nov 24 23:41:06] -- Reloading module 'pbx_dundi.so' (Distributed Universal Number Discovery (DUNDi))
[Nov 24 23:41:06] ERROR[8947]: netsock2.c:305 ast_sockaddr_resolve: getaddrinfo("vicibox10", "(null)", ...): Name or service not known
[Nov 24 23:41:06] WARNING[8947]: acl.c:892 resolve_first: Unable to lookup 'vicibox10'
[Nov 24 23:41:06] WARNING[9077]: pbx_dundi.c:4870 set_config: Unable to look up host 'vicibox10'
[Nov 24 23:41:06] -- Reloading module 'codec_dahdi.so' (Generic DAHDI Transcoder Codec Translator)
[Nov 24 23:41:06] -- Reloading module 'app_alarmreceiver.so' (Alarm Receiver for Asterisk)
[Nov 24 23:41:06] -- Reloading module 'app_followme.so' (Find-Me/Follow-Me Application)
[Nov 24 23:41:06] == Using SIP CoS mark 4

[disepa]

When I do a debug, I get this :
[quote]vicibox10*CLI> sip set debug on
SIP Debugging enabled
[Nov 25 00:11:12] Reliably Transmitting (NAT) to XX.YY.ZZ.AAA:5060:
[Nov 25 00:11:12] OPTIONS sip:xx.sip.callr.xx SIP/2.0
[Nov 25 00:11:12] Via: SIP/2.0/UDP 196.47.175.98:5060;branch=z9hG4bK441e63e4;rport
[Nov 25 00:11:12] Max-Forwards: 70
[Nov 25 00:11:12] From: "asterisk" <sip:asterisk@196.47.175.98>;tag=as2c3e89c8
[Nov 25 00:11:12] To: <sip:xx.sip.callr.xx>
[Nov 25 00:11:12] Contact: <sip:asterisk@196.47.175.98:5060>
[Nov 25 00:11:12] Call-ID: 39e22f8458de9552783ca8265426fa10@196.47.175.98:5060
[Nov 25 00:11:12] CSeq: 102 OPTIONS
[Nov 25 00:11:12] User-Agent: Asterisk PBX 13.38.2-vici
[Nov 25 00:11:12] Date: Wed, 24 Nov 2021 23:11:12 GMT
[Nov 25 00:11:12] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Nov 25 00:11:12] Supported: replaces, timer
[Nov 25 00:11:12] Content-Length: 0
[Nov 25 00:11:12]
[Nov 25 00:11:12]
[Nov 25 00:11:12] ---
[Nov 25 00:11:12] Reliably Transmitting (NAT) to DD.CCC.BBB.YY:5060:
[Nov 25 00:11:12] OPTIONS sip:DD.CCC.BBB.YY SIP/2.0
[Nov 25 00:11:12] Via: SIP/2.0/UDP 196.47.175.98:5060;branch=z9hG4bK48971c42;rport
[Nov 25 00:11:12] Max-Forwards: 70
[Nov 25 00:11:12] From: "asterisk" <sip:asterisk@196.47.175.98>;tag=as6580701b
[Nov 25 00:11:12] To: <sip:5DD.CCC.BBB.YY>
[Nov 25 00:11:12] Contact: <sip:asterisk@196.47.175.98:5060>
[Nov 25 00:11:12] Call-ID: 3229c59c46e2fc5c194dba25202a42e5@196.47.175.98:5060
[Nov 25 00:11:12] CSeq: 102 OPTIONS
[Nov 25 00:11:12] User-Agent: Asterisk PBX 13.38.2-vici
[Nov 25 00:11:12] Date: Wed, 24 Nov 2021 23:11:12 GMT
[Nov 25 00:11:12] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Nov 25 00:11:12] Supported: replaces, timer
[Nov 25 00:11:12] Content-Length: 0

[disepa]

[Nov 25 00:11:12] <--- SIP read from UDP:XX.YY.ZZ.AAA:5060 --->
[Nov 25 00:11:12] SIP/2.0 200 Yo what's up?
[Nov 25 00:11:12] Via: SIP/2.0/UDP 192.168.88.245:5060;branch=z9hG4bK441e63e4;rport=12435;received=196.47.175.98
[Nov 25 00:11:12] From: "asterisk" <sip:asterisk@192.168.88.245:5060>;tag=as2c3e89c8
[Nov 25 00:11:12] To: <sip:xx.sip.callr.xx>;tag=bf8638324618dc61059d4c604476fea1.30f39a63
[Nov 25 00:11:12] Call-ID: 39e22f8458de9552783ca8265426fa10@192.168.2.222:5060
[Nov 25 00:11:12] CSeq: 102 OPTIONS
[Nov 25 00:11:12] Server: K/CALLR 1.42
[Nov 25 00:11:12] Content-Length: 0
[Nov 25 00:11:12]
[Nov 25 00:11:12] <------------->
[Nov 25 00:11:12] --- (8 headers 0 lines) ---
[Nov 25 00:11:12]
[Nov 25 00:11:12] <--- SIP read from UDP:DD.CCC.BBB.YY:5060 --->
[Nov 25 00:11:12] SIP/2.0 200 OK
[Nov 25 00:11:12] Via: SIP/2.0/UDP 192.168.88.245:5060;received=196.47.175.98;branch=z9hG4bK48971c42;rport=12435
[Nov 25 00:11:12] From: "asterisk" <sip:asterisk@192.168.88.245:5060>;tag=as6580701b
[Nov 25 00:11:12] To: <sip:DD.CCC.BBB.YY>;tag=a813c362a81746b22083ddcdd3e8014f.f8ad
[Nov 25 00:11:12] Call-ID: 3229c59c46e2fc5c194dba25202a42e5@192.168.2.222:5060
[Nov 25 00:11:12] CSeq: 102 OPTIONS
[Nov 25 00:11:12] Server: ProvectioVoicePlatform-v2020-PRODUCTION.334
[Nov 25 00:11:12] Content-Length: 0
[Nov 25 00:11:12]

[disepa]

[Nov 25 00:11:12] <------------->
[Nov 25 00:11:12] --- (8 headers 0 lines) ---
[Nov 25 00:11:13] Retransmitting #1 (NAT) to XX.YY.ZZ.AAA:5060:
[Nov 25 00:11:13] OPTIONS sip:xx.sip.callr.xx SIP/2.0
[Nov 25 00:11:13] Via: SIP/2.0/UDP 196.47.175.98:5060;branch=z9hG4bK441e63e4;rport
[Nov 25 00:11:13] Max-Forwards: 70
[Nov 25 00:11:13] From: "asterisk" <sip:asterisk@196.47.175.98>;tag=as2c3e89c8
[Nov 25 00:11:13] To: <sip:xx.sip.callr.xx>
[Nov 25 00:11:13] Contact: <sip:asterisk@196.47.175.98:5060>
[Nov 25 00:11:13] Call-ID: 39e22f8458de9552783ca8265426fa10@196.47.175.98:5060
[Nov 25 00:11:13] CSeq: 102 OPTIONS
[Nov 25 00:11:13] User-Agent: Asterisk PBX 13.38.2-vici
[Nov 25 00:11:13] Date: Wed, 24 Nov 2021 23:11:12 GMT
[Nov 25 00:11:13] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Nov 25 00:11:13] Supported: replaces, timer
[Nov 25 00:11:13] Content-Length: 0
[Nov 25 00:11:13]
[Nov 25 00:11:13]
[Nov 25 00:11:13] ---
[Nov 25 00:11:13] Retransmitting #1 (NAT) to DD.CCC.BBB.YY:5060:
[Nov 25 00:11:13] OPTIONS sip:DD.CCC.BBB.YY SIP/2.0
[Nov 25 00:11:13] Via: SIP/2.0/UDP 196.47.175.98:5060;branch=z9hG4bK48971c42;rport
[Nov 25 00:11:13] Max-Forwards: 70
[Nov 25 00:11:13] From: "asterisk" <sip:asterisk@196.47.175.98>;tag=as6580701b
[Nov 25 00:11:13] To: <sip:DD.CCC.BBB.YY>
[Nov 25 00:11:13] Contact: <sip:asterisk@196.47.175.98:5060>
[Nov 25 00:11:13] Call-ID: 3229c59c46e2fc5c194dba25202a42e5@196.47.175.98:5060
[Nov 25 00:11:13] CSeq: 102 OPTIONS
[Nov 25 00:11:13] User-Agent: Asterisk PBX 13.38.2-vici
[Nov 25 00:11:13] Date: Wed, 24 Nov 2021 23:11:12 GMT
[Nov 25 00:11:13] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Nov 25 00:11:13] Supported: replaces, timer
[Nov 25 00:11:13] Content-Length: 0
[Nov 25 00:11:13]
[Nov 25 00:11:13]
[Nov 25 00:11:13] ---
[Nov 25 00:11:13]
[Nov 25 00:11:13] <--- SIP read from UDP:XX.YY.ZZ.AAA:5060 --->
[Nov 25 00:11:13] SIP/2.0 200 Yo what's up?
[Nov 25 00:11:13] Via: SIP/2.0/UDP 192.168.88.245:5060;branch=z9hG4bK441e63e4;rport=12435;received=196.47.175.98
[Nov 25 00:11:13] From: "asterisk" <sip:asterisk@192.168.88.245:5060>;tag=as2c3e89c8
[Nov 25 00:11:13] To: <sip:xx.sip.callr.xx>;tag=bf8638324618dc61059d4c604476fea1.30f39a63
[Nov 25 00:11:13] Call-ID: 39e22f8458de9552783ca8265426fa10@192.168.2.222:5060
[Nov 25 00:11:13] CSeq: 102 OPTIONS
[Nov 25 00:11:13] Server: K/CALLR 1.42
[Nov 25 00:11:13] Content-Length: 0
[Nov 25 00:11:13]
[Nov 25 00:11:13] <------------->
[Nov 25 00:11:13] --- (8 headers 0 lines) ---
[Nov 25 00:11:13]
[Nov 25 00:11:13] <--- SIP read from UDP:DD.CCC.BBB.YY:5060 --->
[Nov 25 00:11:13] SIP/2.0 200 OK
[Nov 25 00:11:13] Via: SIP/2.0/UDP 192.168.88.245:5060;received=196.47.175.98;branch=z9hG4bK48971c42;rport=12435
[Nov 25 00:11:13] From: "asterisk" <sip:asterisk@192.168.88.245:5060>;tag=as6580701b
[Nov 25 00:11:13] To: <sip:DD.CCC.BBB.YY>;tag=a813c362a81746b22083ddcdd3e8014f.f8ad
[Nov 25 00:11:13] Call-ID: 3229c59c46e2fc5c194dba25202a42e5@192.168.2.222:5060
[Nov 25 00:11:13] CSeq: 102 OPTIONS
[Nov 25 00:11:13] Server: ProvectioVoicePlatform-v2020-PRODUCTION.334
[Nov 25 00:11:13] Content-Length: 0

So what's wrong?

Thanks ad regard

[ambiorixg12]

Your SIP trace is incomplete, you're talking about a REGISTRATION request, but there is none on your SIP trace.

[disepa]

Ok so can you tell me what command line I have to use to have SIP trace please?

Thank you

[ambiorixg12]

You already did it sip set debug on ( but you havent explain what's the issue if is inbound calls issue , using registration or outbound call issue using username and password, note that register string is used for inbound calls, Asterisk tell the provider how to be contacted

[callcentertech]

Your carrier details are okay, looks like the carrier should allow your dialer IP in their firewall.
Pls check with another carrier. Post results here so I can further advise..

Hi everybody,

It's been hard for me because I read any manual on vicidial but my IP trunk by authebtication (CALLR) does not registred.

This is my settings :

[CALLR]
disallow=all
allow=g729
allow=alaw
type=peer
host=xx.sip.callr.xx
dtmfmode=rfc2833
context=default
qualify=yes
nat=force_rport
insecure=port,invite

When I do a SIP show peers :

vicibox10*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
2003/2003 192.168.88.202 D Yes Yes 52916 OK (3 ms)
CALLR XX.YY.ZZ.AAA Yes Yes 5060 UNREACHABLE
KCALL DD.CCC.BBB.YY Yes No 5060 UNREACHABLE
gs102/gs102 (Unspecified) D Yes Yes 0 UNKNOWN