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 Θέμα δημοσίευσης: springtel.com busy διαρκως
ΔημοσίευσηΔημοσιεύτηκε: 03 Οκτ 2013, 09:55 
SIP user

Εγγραφή: 05 Οκτ 2011, 13:26
Δημοσιεύσεις: 55
Καλημέρα σας.
Είχα τον ΟΤΕ isdn και παράλληλα για ένα διάστημα είχα και ένα siptrunk της springtel.com το οποίο λειτούργησε μέχρι πριν φύγω διακοπές. Όταν γύρισα έβαλα forthnet idsn η οποία λειτουργεί πλέον με ακριβώς τις ίδιες ρυθμίσεις στο elastix και προς το παρόν χωρίς προβλήματα.
Το πρόβλημα είναι (δεν παρακολούθησα χρονικά πότε έγινε) ότι δεν μπορώ να βγάλω κλήση από το sip trunk της springtel μου λέει διαρκώς busy.
Μίλησα με το τεχνικό τους τμήμα και μου είπαν


=================
Βλέπουμε ότι το PBX σαςλαμβάνει την απάντηση και απαντάει με ACK (άρα μέχρι εδώ πάμε σωστά και δεν έχουμε κανένα πρόβλημα
επικοινωνίας).
Κατόπιν όμως πρέπει το PBX πρέπει να στείλει ένα νέο INVITE όπου θα έχει απαντήσει και με το password για να γίνει το
authentication

δηλαδή ένα νέο INVITE που να περιλαμβάνει και την απάντηση στο digest authorization.

Authorization: Digest username="ΧΧΧΧΧΧΧΧΧΧΧΧΧΧΧΧΧΧ", realm="194.219.28.67", algorithm=MD5, uri="sip:ΧΧΧΧΧΧΧΧΧΧΧΧ@194.219.28.67", nonce="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXX", response="XXXXXXXXXXXXXXXXXXXXXXX"

Κάτι τέτοιο δεν έρχεται από την πλευρά σας, οπότε πιθανώς κάτι έχει αλλάξει στην εγκατάσταση του Asterisk.

Μπορείτε να ελέγξετε το configuration του PBX σας?
Επί πλέον, καλό θα είναι να προσθέσετε και το G.729 στους acceptable codecs.
Ο,τι άλλη βοήθεια χρειαστείτε είμαστε στη διάθεσή σας
Spring VoIP Operations
=================

άλλαξα τις ρυθμίσεις του trunk από
----------------------------------------------------
type=friend
secret=XXXXXXXXXXXXXX
host=sip.springtel.com
fromuser=0000164355χχχχ
fromdomain=spingtel.com
dtmfmode=inband&inband
disallow=all
allow=g729&ilbc&ulaw&alaw
canreinvite=yes
-----------------------------------------------------
σε
-----------------------------------------------------
type=peer
secret=χχχχχχχχχχχχχχ
host=sip.springtel.com
fromuser=0000164355χχχχ
fromdomain=spingtel.com
dtmfmode=rfc2833
disallow=all
allow=g729&ilbc&ulaw&alaw
canreinvite=yes
-------------------------------------------------

αλλά τα ίδια η κονσόλα μου δίνει το εξής όταν κάνω κλήση με ενεργό μόνο αυτό το trunk
================
-- Registered extension context 'app-recordings'; registrar: pbx_config
-- merging incls/swits/igpats from old(app-recordings) to new(app-recordings ) context, registrar = pbx_ael
-- Including context 'app-recordings-custom' in context 'app-recordings'
-- Added extension '*99' priority 3 to app-recordings
-- Added extension '*99' priority 2 to app-recordings
-- Added extension '*99' priority 1 to app-recordings
-- Added extension '*77' priority 3 to app-recordings
-- Added extension '*77' priority 2 to app-recordings
-- Added extension '*77' priority 1 to app-recordings
-- Registered extension context 'macro-record-enable'; registrar: pbx_config
-- merging incls/swits/igpats from old(macro-record-enable) to new(macro-rec ord-enable) context, registrar = pbx_ael
-- Including context 'macro-record-enable-custom' in context 'macro-record-e nable'
-- Added extension 's' priority 25 to macro-record-enable
-- Added extension 's' priority 24 to macro-record-enable
-- Added extension 's' priority 23 to macro-record-enable
-- Added extension 's' priority 22 to macro-record-enable
-- Added extension 's' priority 21 to macro-record-enable
-- Added extension 's' priority 20 to macro-record-enable
-- Added extension 's' priority 19 to macro-record-enable
-- Added extension 's' priority 18 to macro-record-enable
-- Added extension 's' priority 17 to macro-record-enable
-- Added extension 's' priority 16 to macro-record-enable
-- Added extension 's' priority 15 to macro-record-enable
-- Added extension 's' priority 14 to macro-record-enable
-- Added extension 's' priority 13 to macro-record-enable
-- Added extension 's' priority 12 to macro-record-enable
-- Added extension 's' priority 11 to macro-record-enable
-- Added extension 's' priority 10 to macro-record-enable
-- Added extension 's' priority 9 to macro-record-enable
-- Added extension 's' priority 8 to macro-record-enable
-- Added extension 's' priority 7 to macro-record-enable
-- Added extension 's' priority 6 to macro-record-enable
-- Added extension 's' priority 5 to macro-record-enable
-- Added extension 's' priority 4 to macro-record-enable
-- Added extension 's' priority 3 to macro-record-enable
-- Added extension 's' priority 2 to macro-record-enable
-- Added extension 's' priority 1 to macro-record-enable
-- Registered extension context 'macro-hangupcall'; registrar: pbx_config
-- merging incls/swits/igpats from old(macro-hangupcall) to new(macro-hangup call) context, registrar = pbx_ael
-- Including context 'macro-hangupcall-custom' in context 'macro-hangupcall'
-- Added extension 's' priority 34 to macro-hangupcall
-- Added extension 's' priority 33 to macro-hangupcall
-- Added extension 's' priority 32 to macro-hangupcall
-- Added extension 's' priority 31 to macro-hangupcall
-- Added extension 's' priority 30 to macro-hangupcall
-- Added extension 's' priority 29 to macro-hangupcall
-- Added extension 's' priority 28 to macro-hangupcall
-- Added extension 's' priority 27 to macro-hangupcall
-- Added extension 's' priority 26 to macro-hangupcall
-- Added extension 's' priority 25 to macro-hangupcall
-- Added extension 's' priority 24 to macro-hangupcall
-- Added extension 's' priority 23 to macro-hangupcall
-- Added extension 's' priority 22 to macro-hangupcall
-- Added extension 's' priority 21 to macro-hangupcall
-- Added extension 's' priority 20 to macro-hangupcall
-- Added extension 's' priority 19 to macro-hangupcall
-- Added extension 's' priority 18 to macro-hangupcall
-- Added extension 's' priority 17 to macro-hangupcall
-- Added extension 's' priority 16 to macro-hangupcall
-- Added extension 's' priority 15 to macro-hangupcall
-- Added extension 's' priority 14 to macro-hangupcall
-- Added extension 's' priority 13 to macro-hangupcall
-- Added extension 's' priority 12 to macro-hangupcall
-- Added extension 's' priority 11 to macro-hangupcall
-- Added extension 's' priority 10 to macro-hangupcall
-- Added extension 's' priority 9 to macro-hangupcall
-- Added extension 's' priority 8 to macro-hangupcall
-- Added extension 's' priority 7 to macro-hangupcall
-- Added extension 's' priority 6 to macro-hangupcall
-- Added extension 's' priority 5 to macro-hangupcall
-- Added extension 's' priority 4 to macro-hangupcall
-- Added extension 's' priority 3 to macro-hangupcall
-- Added extension 's' priority 2 to macro-hangupcall
-- Added extension 's' priority 1 to macro-hangupcall
-- Time to scan old dialplan and merge leftovers back into the new: 0.022041 sec
-- Time to restore hints and swap in new dialplan: 0.000007 sec
-- Time to delete the old dialplan: 0.000852 sec
-- Total time merge_contexts_delete: 0.022900 sec
-- Reloading module 'app_amd.so' (Answering Machine Detection Application)
-- Reloading module 'chan_agent.so' (Agent Proxy Channel)
-- Reloading module 'app_playback.so' (Sound File Playback Application)
-- Reloading module 'cdr_odbc.so' (ODBC CDR Backend)
-- Reloading module 'chan_dahdi.so' (DAHDI Telephony Driver w/PRI & MFC/R2)
== Parsing '/etc/asterisk/chan_dahdi.conf': == Found
== Parsing '/etc/asterisk/dahdi-channels.conf': == Found
== Parsing '/etc/asterisk/chan_dahdi_additional.conf': == Found
== Parsing '/etc/asterisk/users.conf': == Found
-- Reconfigured channel 1, ISDN BRI Point to Point signalling
-- Reconfigured channel 2, ISDN BRI Point to Point signalling
-- Reloading module 'cel_manager.so' (Asterisk Manager Interface CEL Backend )
-- Reloading module 'codec_alaw.so' (A-law Coder/Decoder)
-- Reloading module 'cdr_sqlite3_custom.so' (SQLite3 Custom CDR Module)
-- Reloading module 'app_queue.so' (True Call Queueing)
== Parsing '/etc/asterisk/queues.conf': == Found
== Parsing '/etc/asterisk/queues_general_additional.conf': == Found
== Parsing '/etc/asterisk/queues_custom_general.conf': == Found
== Parsing '/etc/asterisk/queues_custom.conf': == Found
== Parsing '/etc/asterisk/queues_additional.conf': == Found
== Parsing '/etc/asterisk/queues_post_custom.conf': == Found
-- Reloading module 'res_config_curl.so' (Realtime Curl configuration)
== Parsing '/etc/asterisk/res_curl.conf': == Found
-- Reloading module 'res_clialiases.so' (CLI Aliases)
-- Reloading module 'codec_adpcm.so' (Adaptive Differential PCM Coder/Decode r)
-- Reloading module 'cel_odbc.so' (ODBC CEL backend)
== Parsing '/etc/asterisk/cel_odbc.conf': == Found
-- Reloading module 'app_voicemail.so' (Comedian Mail (Voicemail System))
== Parsing '/etc/asterisk/voicemail.conf': == Found
== Parsing '/etc/asterisk/vm_general.inc': == Found
== Parsing '/etc/asterisk/vm_email.inc': == Found
== Parsing '/etc/asterisk/users.conf': == Found
-- Reloading module 'cel_pgsql.so' (PostgreSQL CEL Backend)
-- Reloading module 'chan_mgcp.so' (Media Gateway Control Protocol (MGCP))
Reloading MGCP
-- Reloading module 'cel_custom.so' (Customizable Comma Separated Values CEL Backend)
== Parsing '/etc/asterisk/cel_custom.conf': == Found
-- Reloading module 'pbx_dundi.so' (Distributed Universal Number Discovery ( DUNDi))
== Parsing '/etc/asterisk/dundi.conf': == Found
== Parsing '/etc/asterisk/dundi_general_custom_elastix.conf': == Found
== Parsing '/etc/asterisk/dundi_mappings_custom_elastix.conf': == Found
== Parsing '/etc/asterisk/dundi_peers_custom_elastix.conf': == Found
== Parsing '/etc/asterisk/sip_notify.conf': == Found
== Parsing '/etc/asterisk/sip_notify_custom.conf': == Found
== Parsing '/etc/asterisk/sip_notify_custom_elastix.conf': == Found
== Parsing '/etc/asterisk/sip_notify_additional.conf': == Found
== Parsing '/etc/asterisk/res_pktccops.conf': == Found
webintel*CLI>
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [2103463933@from-internal:1] Macro("SIP/10-0000039e", "user-callerid,SKIPTTL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/10-0000039e", "AMPUSER=10") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/10-0000039e", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/10-0000039e", "1?Set(REALCALLERIDNUM=10)") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/10-0000039e", "AMPUSER=10") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/10-0000039e", "AMPUSERCIDNAME=Cris") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/10-0000039e", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/10-0000039e", "AMPUSERCID=10") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/10-0000039e", "CALLERID(all)="Cris" <10>") in new stack
-- Executing [s@macro-user-callerid:9] ExecIf("SIP/10-0000039e", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:10] GotoIf("SIP/10-0000039e", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] Set("SIP/10-0000039e", "CALLERID(number)=10") in new stack
-- Executing [s@macro-user-callerid:20] Set("SIP/10-0000039e", "CALLERID(name)=Cris") in new stack
-- Executing [s@macro-user-callerid:21] NoOp("SIP/10-0000039e", "Using CallerID "Cris" <10>") in new stack
-- Executing [2103463933@from-internal:2] NoOp("SIP/10-0000039e", "Calling Out Route: 9_outside") in new stack
-- Executing [2103463933@from-internal:3] Set("SIP/10-0000039e", "MOHCLASS=default") in new stack
-- Executing [2103463933@from-internal:4] Set("SIP/10-0000039e", "_NODEST=") in new stack
-- Executing [2103463933@from-internal:5] Macro("SIP/10-0000039e", "record-enable,10,OUT,") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/10-0000039e", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] ExecIf("SIP/10-0000039e", "0?MacroExit()") in new stack
-- Executing [s@macro-record-enable:5] GotoIf("SIP/10-0000039e", "0?Group:OUT") in new stack
-- Goto (macro-record-enable,s,15)
-- Executing [s@macro-record-enable:15] GotoIf("SIP/10-0000039e", "0?IN") in new stack
-- Executing [s@macro-record-enable:16] ExecIf("SIP/10-0000039e", "1?MacroExit()") in new stack
-- Executing [2103463933@from-internal:6] Macro("SIP/10-0000039e", "dialout-trunk,1,2103463933,") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/10-0000039e", "DIAL_TRUNK=1") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/10-0000039e", "0?sub-pincheck,s,1") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/10-0000039e", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/10-0000039e", "DIAL_NUMBER=2103463933") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/10-0000039e", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/10-0000039e", "OUTBOUND_GROUP=OUT_1") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/10-0000039e", "0?nomax") in new stack
-- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/10-0000039e", "0?chanfull") in new stack
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/10-0000039e", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/10-0000039e", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/10-0000039e", "outbound-callerid,1") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/10-0000039e", "0?Set(CALLERPRES()=)") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/10-0000039e", "0?Set(REALCALLERIDNUM=10)") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/10-0000039e", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/10-0000039e", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/10-0000039e", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/10-0000039e", "TRUNKOUTCID=2114043815") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/10-0000039e", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/10-0000039e", "1?Set(CALLERID(all)=2114043815)") in new stack
-- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/10-0000039e", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/10-0000039e", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/10-0000039e", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
-- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/10-0000039e", "0?sub-flp-1,s,1") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/10-0000039e", "OUTNUM=2103463933") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/10-0000039e", "custom=SIP/spring") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/10-0000039e", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
-- Executing [s@macro-dialout-trunk:16] Macro("SIP/10-0000039e", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/10-0000039e", "") in new stack
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/10-0000039e", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/10-0000039e", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/10-0000039e", "SIP/spring/2103463933,300,") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/spring/2103463933
-- SIP/spring-0000039f is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:20] NoOp("SIP/10-0000039e", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 21") in new stack
-- Executing [s@macro-dialout-trunk:21] Goto("SIP/10-0000039e", "s-CONGESTION,1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("SIP/10-0000039e", "RC=21") in new stack
-- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/10-0000039e", "21,1") in new stack
-- Goto (macro-dialout-trunk,21,1)
-- Executing [21@macro-dialout-trunk:1] Goto("SIP/10-0000039e", "continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/10-0000039e", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,continue,3)
-- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/10-0000039e", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 21 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:4] Set("SIP/10-0000039e", "CALLERID(number)=10") in new stack
-- Executing [2103463933@from-internal:7] Macro("SIP/10-0000039e", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/10-0000039e", "") in new stack
-- Executing [s@macro-outisbusy:2] GotoIf("SIP/10-0000039e", "0?emergency,1") in new stack
-- Executing [s@macro-outisbusy:3] GotoIf("SIP/10-0000039e", "0?intracompany,1") in new stack
-- Executing [s@macro-outisbusy:4] Playback("SIP/10-0000039e", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
-- <SIP/10-0000039e> Playing 'all-circuits-busy-now.gsm' (language 'en')
-- <SIP/10-0000039e> Playing 'pls-try-call-later.gsm' (language 'en')
webintel*CLI>
=================

Μίλησα μαζί τους ξανά τους έστειλα τα αποτελέσματα και μου απάντησαν αυτό
=================
Από την πλευρά μας βλέπουμε
ότι δεν υπάρχει κάποιο πρόβλημα με τον Spring VoIP λογαριασμό σας και το πρόβλημα είναι ότι το PBX σας
δεν στέλνει digest authorization στο INVITE, όπως σας έχουμε εξηγήσει σε προηγούμενο email.
==================

Ευχαριστώ για την βοήθεια.


Κορυφή
Χωρίς σύνδεση Προφίλ  
 
 Θέμα δημοσίευσης: Re: springtel.com busy διαρκως
ΔημοσίευσηΔημοσιεύτηκε: 03 Οκτ 2013, 11:13 
SIP debugger

Εγγραφή: 26 Μαρ 2009, 20:02
Δημοσιεύσεις: 1007
Δοκιμάστε τα εξής

type=friend
username=0000164355χχχχ
secret=XXXXXXXXXXXXXX
host=sip.springtel.com
fromdomain=spingtel.com
dtmfmode=inband&inband
disallow=all
allow=g729&ilbc&ulaw&alaw
insecure=very
canreinvite=no


Κορυφή
Χωρίς σύνδεση Προφίλ  
 
 Θέμα δημοσίευσης: Re: springtel.com busy διαρκως
ΔημοσίευσηΔημοσιεύτηκε: 03 Οκτ 2013, 11:53 
SIP user

Εγγραφή: 05 Οκτ 2011, 13:26
Δημοσιεύσεις: 55
Τι να πω!
Ευχαριστώ πολύ.


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Δημιουργία νέου θέματος Απαντήστε στο θέμα  [ 3 Δημοσιεύσεις ]  Συντονιστής: allVoIP team

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