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Bekijk Volledige Versie : Probleem met Asterisk/Trixbox (kan soms niet gebeld worden)



Rick L.
26/06/06, 11:18
Goedendag allen,

Ik ben de laatste weken aan het experimenteren geweest met Trixbox om een telefooncentrale aan de praat te krijgen. Dit werkt aardig, maar 2 van de 10 keer pakt Asterisk het signaal niet en doet er niets mee, waardoor de beller geen verbinding krijgt. Ik heb deze vraag al op diverse forums gezet maar heb nergens een reactie gekregen, dus ik weet niet goed waar ik hulp kan vinden. Ik hoop dat hier iemand is die het probleem bekend voorkomt of tips heeft voor het verhelpen ervan.

Ik gebruik op dit moment VoipBuster voor inkomende en uitgaande lijnen. Asterisk is op dit moment heel simpel ingesteld: Als iemand belt gaat Ring Group 1 over, met daaraan één telefoon gekoppeld. Als deze telefoon in gesprek is of niet bereikbaar is, neemt IVR hem over. Acht van de tien keer werkt het vlekkeloos, maar die andere twee keer toch niet. En waar het probleem nou in zit... ik heb geen flauw idee.

Wanneer ik bel en de telefoon niet over gaat, lijkt het signaal wel binnen te komen op de server:


Jun 24 10:47:03 DEBUG[5434] manager.c: Manager received command 'Command'
Jun 24 10:47:03 DEBUG[5434] manager.c: Manager received command 'Command'
Jun 24 10:47:04 DEBUG[2509] chan_sip.c: Allocating new SIP dialog for 5e74af81386240f1a305a109b5448a15 - INVITE (With RTP)
Jun 24 10:47:04 DEBUG[2509] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
Jun 24 10:47:04 DEBUG[2509] chan_sip.c: Setting NAT on RTP to 0
Jun 24 10:47:04 DEBUG[2509] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
Jun 24 10:47:04 DEBUG[2509] chan_sip.c: Stopping retransmission on '5e74af81386240f1a305a109b5448a15' of Response 4: Match Found
Jun 24 10:47:07 DEBUG[2509] chan_sip.c: Allocating new SIP dialog for 471619a9deda487c88a82b3329ae9d05 - INVITE (With RTP)
Jun 24 10:47:07 DEBUG[2509] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
Jun 24 10:47:07 DEBUG[2509] chan_sip.c: Setting NAT on RTP to 0
Jun 24 10:47:07 DEBUG[2509] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
Jun 24 10:47:07 DEBUG[2509] chan_sip.c: Stopping retransmission on '471619a9deda487c88a82b3329ae9d05' of Response 6: Match Found
Jun 24 10:47:13 DEBUG[2509] chan_sip.c: Allocating new SIP dialog for 444fffc39fc94cdcb36d79e4443cf6c0 - INVITE (With RTP)
Jun 24 10:47:13 DEBUG[2509] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
Jun 24 10:47:13 DEBUG[2509] chan_sip.c: Setting NAT on RTP to 0
Jun 24 10:47:13 DEBUG[2509] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
Jun 24 10:47:13 DEBUG[2509] chan_sip.c: Stopping retransmission on '444fffc39fc94cdcb36d79e4443cf6c0' of Response 8: Match Found
Jun 24 10:47:16 DEBUG[2509] chan_sip.c: Allocating new SIP dialog for 35f9106d2068489fa8c164b9f52d0a79 - INVITE (With RTP)
Jun 24 10:47:16 DEBUG[2509] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
Jun 24 10:47:16 DEBUG[2509] chan_sip.c: Setting NAT on RTP to 0
Jun 24 10:47:16 DEBUG[2509] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
Jun 24 10:47:16 DEBUG[2509] chan_sip.c: Stopping retransmission on '35f9106d2068489fa8c164b9f52d0a79' of Response 10: Match Found

En als ik ophang:


Jun 24 10:47:19 DEBUG[2509] chan_sip.c: Auto destroying call '5e74af81386240f1a305a109b5448a15'
Jun 24 10:47:22 DEBUG[2509] chan_sip.c: Auto destroying call '471619a9deda487c88a82b3329ae9d05'
Jun 24 10:47:28 DEBUG[2509] chan_sip.c: Auto destroying call '444fffc39fc94cdcb36d79e4443cf6c0'
Jun 24 10:47:31 DEBUG[2509] chan_sip.c: Auto destroying call '35f9106d2068489fa8c164b9f52d0a79'
Jun 24 10:47:35 DEBUG[2509] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP)
Jun 24 10:47:35 DEBUG[2509] chan_sip.c: Stopping retransmission on '7d65923c3367267f606426bf55b7e1b4@62.212.*.*' of Request 102: Match Found

Daarvoor verschijnen geet foutmeldingen over dat Asterisk niet kan connecten met Voipbuster. Integendeel, elke twee minuten verbindt hij zonder problemen en geeft altijd de melding "Registration succesful":


Jun 24 20:15:12 DEBUG[12098] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP)
Jun 24 20:15:12 DEBUG[12098] chan_sip.c: Stopping retransmission on '78d3e5091525e3f75e7eb49a4f318149@62.212.*.*' of Request 102: Match Found
Jun 24 20:16:12 DEBUG[12098] chan_sip.c: Allocating new SIP dialog for 16064644349841a70815450b666bcbc7@127.0.0.1 - REGISTER (No RTP)
Jun 24 20:16:12 DEBUG[12098] chan_sip.c: Scheduled a registration timeout for sip1.voipbuster.com id #1838
Jun 24 20:16:12 VERBOSE[12098] logger.c: REGISTER attempt 1 to myusername@sip1.voipbuster.com
Jun 24 20:16:12 DEBUG[12098] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '16064644349841a70815450b666bcbc7@127.0.0.1' Request 223: Found
Jun 24 20:16:12 DEBUG[12098] chan_sip.c: Stopping retransmission on '16064644349841a70815450b666bcbc7@127.0.0.1' of Request 223: Match Found
Jun 24 20:16:12 DEBUG[12098] chan_sip.c: Registration successful
Jun 24 20:16:12 DEBUG[12098] chan_sip.c: Cancelling timeout 1838
Jun 24 20:16:12 DEBUG[12098] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP)
Jun 24 20:16:12 DEBUG[12098] chan_sip.c: Stopping retransmission on '41148fb0654d25305242583d1e5e0574@62.212.*.*' of Request 102: Match Found
Jun 24 20:16:14 DEBUG[12478] manager.c: Manager received command 'Command'
Jun 24 20:16:14 DEBUG[12478] manager.c: Manager received command 'Command'

Toch heb ik het vermoeden dat het ergens om een instelling bij de Trunks moet zijn. Ik gebruik op dit moment de volgende gegevens:

Outgoing Settings

Trunk Name: VoipBuster

Peer details:


allow=alaw
canreinvite=no
disallow=all
dtmfmode=inband
fromdomain=sip.voipbuster.com
fromuser=mijngebruikersnaam
host=sip.voipbuster.com
qualify=yes
secret=*****
type=peer
username=mijngebruikersnaam

INCOMMING SETTINGS

USER Context: from-pstn

USER Details:


canreinvite=no
context=from-trunk
fromuser=mijngebruikersnaam
qualify=no
secret=*****
type=peer
username=mijngebruikersnaam

Ik hoop dat iemand een oplossing of een tip heeft, alles is in elk geval welkom. Ook als iemand hetzelfde probleem heeft en nog geen oplossing gevonden heeft, dat weet ik in ieder geval dat ik niet de enige ben ;) Heb in diverse forums gezocht maar heb nergens iemand kunnen vinden met hetzelfde probleem...

Alvast bedankt!!

Wat mij nu overigens opvalt in de logs is dat hij soms NAT op 0 zet en dat dán de telefoon niet gaat rinkelen (maar ook IVR het signaal niet oppakt):


Jun 26 10:39:39 DEBUG[12098] chan_sip.c: Allocating new SIP dialog for e8f7673523c942e298435d1d1e39d3e2 - INVITE (With RTP)
Jun 26 10:39:39 DEBUG[12098] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
Jun 26 10:39:39 DEBUG[12098] chan_sip.c: Setting NAT on RTP to 0
Jun 26 10:39:39 DEBUG[12098] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
Jun 26 10:39:39 DEBUG[12098] chan_sip.c: Stopping retransmission on 'e8f7673523c942e298435d1d1e39d3e2' of Response 0: Match Found
Jun 26 10:39:42 DEBUG[12098] chan_sip.c: Allocating new SIP dialog for 83167f500a464ab9b524e2b598ed8ce7 - INVITE (With RTP)
Jun 26 10:39:42 DEBUG[12098] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
Jun 26 10:39:42 DEBUG[12098] chan_sip.c: Setting NAT on RTP to 0
Jun 26 10:39:42 DEBUG[12098] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
Jun 26 10:39:42 DEBUG[12098] chan_sip.c: Stopping retransmission on '83167f500a464ab9b524e2b598ed8ce7' of Response 2: Match Found
Jun 26 10:39:43 DEBUG[13103] manager.c: Manager received command 'Command'
Jun 26 10:39:43 DEBUG[13103] manager.c: Manager received command 'Command'
Jun 26 10:39:47 DEBUG[12098] chan_sip.c: Allocating new SIP dialog for 94aa913944f34a258c74adc1b05b1775 - INVITE (With RTP)
Jun 26 10:39:47 DEBUG[12098] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
Jun 26 10:39:47 DEBUG[12098] chan_sip.c: Setting NAT on RTP to 0
Jun 26 10:39:47 DEBUG[12098] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
Jun 26 10:39:47 DEBUG[12098] chan_sip.c: Stopping retransmission on '94aa913944f34a258c74adc1b05b1775' of Response 4: Match Found
Jun 26 10:39:54 DEBUG[12098] chan_sip.c: Auto destroying call 'e8f7673523c942e298435d1d1e39d3e2'
Jun 26 10:39:57 DEBUG[12098] chan_sip.c: Auto destroying call '83167f500a464ab9b524e2b598ed8ce7'
Jun 26 10:40:02 DEBUG[12098] chan_sip.c: Auto destroying call '94aa913944f34a258c74adc1b05b1775'

En als hij wel gaat rinkelen zet hij NAT niet op 0:


Jun 26 10:40:58 DEBUG[12098] chan_sip.c: Allocating new SIP dialog for a530bdc98666465298893a8f4b58121e - INVITE (With RTP)
Jun 26 10:40:58 DEBUG[12098] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
Jun 26 10:40:58 DEBUG[12098] chan_sip.c: Checking SIP call limits for device
Jun 26 10:40:58 DEBUG[12098] chan_sip.c: Updating call counter for incoming call
Jun 26 10:40:58 DEBUG[12098] chan_sip.c: build_route: Contact hop: sip:00@194.120.0.202:5060
Jun 26 10:40:58 DEBUG[12088] chan_sip.c: Checking device state for peer 62.212.*.*
Jun 26 10:40:58 DEBUG[12088] devicestate.c: Changing state for SIP/62.212.*.* - state 2 (In use)
Jun 26 10:40:58 DEBUG[3842] pbx.c: Expression result is '1'
Jun 26 10:40:58 DEBUG[3842] pbx.c: Launching 'GotoIf'
Jun 26 10:40:58 VERBOSE[3842] logger.c: -- Executing GotoIf("SIP/62.212.*.*-08ae2c10", "1?from-trunk||1") in new stack
Jun 26 10:40:58 VERBOSE[3842] logger.c: -- Goto (from-trunk,s,1)
Jun 26 10:40:58 DEBUG[3842] pbx.c: Launching 'Set'
Jun 26 10:40:58 VERBOSE[3842] logger.c: -- Executing Set("SIP/62.212.*.*-08ae2c10", "FROM_DID=s") in new stack
Jun 26 10:40:58 DEBUG[3842] pbx.c: Launching 'Set'

De server staat in een datacenter, dus wellicht dat daar iets mis gaat. Hopelijk kan iemand hier iets mee?